RTMT
Considerations

Security
Considerations

Serviceability Considerations

Regenerate
Certificate

You can
regenerate certificates from the Cisco Unified Communications Operating System
as an operating system security function. For more information about
regenerating certificates, see the
Cisco Unified Communications Manager Security
Guide.

Caution

Regenerating a
certificate can affect your system operations. Regenerating a certificate
overwrites the existing certificate including a third party signed certificate
if one was uploaded.

Note

Certificate
regeneration or upload a of third party signed certificates should be performed
during maintenance.

The following
table describes the system security certificates you can regenerate from the
Cisco Unified Communications Operating System and the related services that
must be restarted. For information about regenerating the TFTP certificate, see
the
Cisco Unified Communications Manager Security
Guide.

Table 1 Certificate
Names and Descriptions

Name

Description

Related
Services

tomcat

This self-signed root certificate is generated during
installation for the HTTPS node.

tomcat

ipsec

This self-signed root certificate is generated during
installation for IPsec connections with MGCP and H.323 gateways.

Cisco
Disaster Recovery System (DRS) Local and Cisco DRF Master

CallManager

This self-signed root certificate is installed automatically
when you install
Cisco Unified
Communications Manager. This certificate provides node
identification, including the node name and the Global Unique Identifier
(GUID).

CallManager and CAPF

CAPF

The system copies this root certificate to your node or to all
nodes in the cluster after you complete the Cisco client configuration.

CallManager and CAPF

TVS

This is a self-signed root certificate.

TVS

If you regenerated
the certificate for Cisco Certificate Authority Proxy Function (CAPF) or Cisco
Unified Communications Manager and a CTL client is configured, rerun the CTL
client.

After you
regenerate certificates in the
Cisco Unified
Communications Operating System, you must perform a system backup so
that the latest backup contains the regenerated certificates. If your backup
does not contain the regenerated certificates and you perform a system
restoration task, you must manually unlock each phone in your system so that
the phone can register with
Cisco Unified
Communications Manager. For information about performing a backup,
see the
Disaster
Recovery System Administration Guide.

Procedure

Step 1

Navigate to
Security > Certificate
Management.

The Certificate
List window displays.

Step 2

Click
Generate
New.

The Generate
Certificate dialog box opens.

Step 3

From the
Certificate Name drop-down list, choose a certificate name .
For details
about certificate names, see the Certificate Names and Descriptions table.

Step 4

From the Key
Length drop-down list, choose 1024 or 2048.

Step 5

From the Hash
Algorithm drop-down list, choose SHA1 or SHA256.

Step 6

Click
Generate
New.

What to Do Next

Restart all
services that are affected by the regenerated certificate as listed in the
Certificate Names and Descriptions table.

Rerun the CTL
client (if configured) after you regenerate the CAPF or CallManager
certificates.

Perform a system
backup to capture the newly regenerated certificates. For information about
performing a backup, see the
Disaster Recovery System
Administration Guide.

Generate Certificate
Signing Request

To
generate a CSR, follow these steps:

Procedure

Step 1

Navigate to
Security > Certificate
Management.

The Certificate
List window displays.

Step 2

Click
Generate
CSR.

The Generate
Certificate Signing Request dialog box opens.

Step 3

From the
Certificate Name drop-down list, choose a certificate name.

For details
about certificate names, see the Certificate Names and Descriptions table.

Step 4

From the Key
Length drop-down list, choose 1024 or 2048.

Step 5

From the Hash
Algorithm drop-down list, choose SHA1 or SHA256.

Step 6

Click
Generate
CSR.

Note

Generating CSR
overwrites any existing CSR.

Acknowledgment in
AuditLog

Cisco Unified
Communications Manager Administration Considerations

You can set up a
warning message when an administrator attempts to sign in to any of the
following
Cisco Unified Communications Manager interfaces:

Cisco Unified
Reporting

Cisco Unified
Communications Manager Administration

Disaster
Recovery System

Cisco Unified
Serviceability

Cisco Unified
Operating System Administration

The administrator
can sign in only after acknowledging the warning message. The acknowledgment is
recorded in the audit logs along with the username of the administrator.

Note

You can enable
the acknowledgment by checking the Require User Acknowledgment checkbox in the
Customized Logon Message window (Software
Upgrades > Customized Logon Message) in the Cisco
Unified Operating System Administrative interface.

Security
Considerations

Serviceability
Considerations

Display AuditApp Logs

From the Select a Node drop-down list, choose the server on which the logs that you want to view are stored.

Step 3

Double-click the AuditApp Logs folder.

Step 4

Click the .log file located outside the Archive folder to view the current logs. The AuditApp Logs for the selected node are displayed in a tabular form.

Note

If you want see the old logs, double-click the Archive folder and click the corresponding file.

Step 5

Double-click the entry that you want to view. The auditlog message for that particular entry appears in a new window.

Tip

You can filter the auditlog message display results by choosing an option in the Filter By drop-down list box. To remove the filter, click Clear Filter. All logs appear after you clear the filter.

Display Cisco Unified OS Logs

Procedure

Step 1

Choose System > Tools > AuditLog Viewer

Step 2

From the Select a Node drop-down list, choose the node where the logs that you want to view are stored.

Step 3

Double-click the Cisco Unified OS Logs folder.

Step 4

Click the vos-audit.log file located outside the Archive folder to view the current logs. The Cisco Unified OS Logs for the selected node appear in a tabular form.

Note

If you want see the old logs, double-click the Archive folder and click the corresponding file.

Step 5

Double-click the entry that you want to view. The Cisco Unified OS log message for that particular entry is displayed in a new window.

Tip

You can filter the Cisco Unified OS log message display results by choosing the set of options in a pop up window that appears after you click Filter. To remove the filter, click Clear Filter. All logs appear after you clear the filter.

Call
Dusting

Call Dusting allows you to transfer an active or hold call session from a hardware endpoint to a support bring your own device (BYOD). You can trigger this feature by pressing a MOVE softkey.

Consider the following call flow:

When the BYOD and hardware endpoint are within proximity of each other, you press a move softkey which triggers Unified Communications Manager to ring all shared-line devices with the same user ID.

After the BYOD answers, Unified Communications Manager seamlessly switches the call from the endpoint to the answering device (BYOD).

The move softkey rings all shared-line devices, whether they are configured as a mobile phone or Remote Destination.

For a dual-mode device that uses single registration, the call rings on the preferred side (Wi-Fi or cellular) which is indicated in the REGISTER message. For other dual-mode clients, the call rings through Wi-Fi first. If the device is not registered to Wi-Fi, the call routes through the cellular network.

Note

When the call is on hold, you can still move the call to BYOD by pressing the move softkey. When the call is dusted to BYOD, the call is active after it is answered.

This feature applies only to supported SIP phones.

Call dusting has higher priority than Time-of-Day Routing, Access List, and Do Not Disturb.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Call Home
Configuration

Cisco Unified
Communications Manager Administration Considerations

The Call Home
feature allows to communicate and send the diagnostic alerts, inventory, and
other messages to the Smart Call Home back-end server.

You can also
configure the Cisco Smart Call Home while installing the Cisco Unified
Communications Manager. During installation, the user can select one of the
following options:

Enabled (Smart Call Home)

Enabled (Anonymous Call
Home)

None

Disabled

The following
table describes the settings to configure the Unified Communications Manager
Call Home.

Table 3 Call Home
Configuration Settings

Field Name

Description

Call Home Message Schedule

Displays the date and time of the last Call Home messages that
were sent and the next message that is scheduled.

Call Home*

From the
drop-down list, select one of the following options:

None: Select this option if you chose the Remind Me Later option during
Cisco Unified Communications Manager installation. A message displays that the
Smart Call Home is not configured during your next login to the Cisco Unified
Serviceability console.

Disabled: Select this option if you disabled the Smart Call Home
functionality during installation.

Enabled (Smart Call Home): Select this option if you selected Smart Call
Home option during installation. By default, this option is enabled during
Cisco Unified Communications Manager installation.

Note

The
values specified during the installation appear in the Cisco Unified
Serviceability console.

Enabled (Anonymous Call Home): Select this option if you selected Anonymous
Call Home option during Cisco Unified Communications Manager installation. When
you select this option, Customer Contact Details is disabled and Send data
section is enabled on call home page.
The
following are the characteristics of Anonymous Call Home:

When you select Anonymous
Call Home, this option sends the System configuration (hardware/VM, CPU) and
Software configuration related information to Cisco Smart Call Home for
information-gathering purposes and to make the product better.

Set Up Cisco
Unified Communications Manager Publisher Node

Follow
this procedure to configure the first server where you install
Cisco
Unified Communications Manager software as the publisher node for the
cluster. Perform this procedure after you have completed the basic installation
and configured the basic installation.

Note

You can
configure Smart Call Home on the publisher node only. For more information on
Smart Call Home, refer to Smart call home section in the Cisco Unified
Serviceability Administration Guide.

Procedure

Step 1

The
Network
Time Protocol Client Configuration window appears.

Cisco recommends
that you use an external NTP server to ensure accurate system time on the
publisher node. Subscriber nodes in the cluster will get their time from the
first node.

Step 2

Choose whether
you want to configure an external NTP server or manually configure the system
time.

To set up an
external NTP server, choose Yes and enter the IP address, NTP server name, or
NTP server pool name for at least one NTP server. You can configure up to five
NTP servers, and Cisco recommends that you use at least three. Choose Proceed
to continue with the installation.
The system
contacts an NTP server and automatically sets the time on the hardware clock.

Note

If the
Test button appears, you can choose Test to check
whether the NTP servers are accessible.

To manually
configure the system time, choose
No and enter the appropriate date and time to set
the hardware clock. Choose
OK to continue with the installation.

The
Database Access Security Configuration window
appears.

Step 3

Enter the
Security password from Required Installation Information.

Note

The Security
password must start with an alphanumeric character, be at least six characters
long, and can contain alphanumeric characters, hyphens, and underscores. The
system uses this password to authorize communications between nodes, and you
must ensure this password is identical on all nodes in the cluster.

The
SMTP
Host Configuration window appears.

Step 4

If you want to
configure an SMTP server, choose
Yes and enter the SMTP server name. If you do not
want to configure the SMTP server, choose
No, which redirects to Smart Call Home page. To go
to previous page, choose
Back and to see the information about the SMTP
configuration, choose
Help.

Note

You must
configure an SMTP server to use certain platform features; however, you can
also configure an SMTP server later by using the platform GUI or the command
line interface.

Step 5

Choose
OK. The
Smart
Call Home Enable window appears.

Step 6

On the Smart
Call Home Enable Page, perform one of the following.

Select
Enable Smart
Call Home on System Start to enable the Call Home, and then click
OK. The Smart Call Home Configuration window
appears.

Select
the method for sending data to the Cisco Technical Assistance Center.

Secure Web (HTTPS)

Secure Web (HTTPS) through Proxy
Enter the Hostname/IP Address and port number for Proxy

Hostname/IP Address—Enter the IP address or the hostname of the
proxy server to send the Call Home messages through an indirect network
connection.

Port—Enter the port number on which the proxy server is enabled.

Email

Note

You must have configured the SMTP for Email to be sent successfully.

To
send a copy of the Call Home messages to multiple email recipients, enter the
email addresses separated with a comma. You can enter up to a maximum of 1024
characters.

Enter
the email address of the customer in the Customer Contact Details field.

Click
Continue to proceed, or select
Back to return to the previous menu. If you click
Continue, a message appears as
Cisco Call Home includes
reporting capabilities that allow Cisco to receive diagnostic and system
information from your
Unified Communications Manager cluster. Cisco may use this
information for proactive debugging, product development or marketing purposes.
To learn more about this feature, please visit:
http://www.cisco.com/en/US/products/ps7334/serv_home.html.

Click
Confirm to proceeds with normal installation or
select
Back to return to the Smart Call Home Enable Page.

Select
Enable
Anonymous Call Home on System Start to enable the Anonymous Call Home, and
then click
OK. The Anonymous Call Home Configuration window
appears.

Select
the method for sending data to the Cisco Technical Assistance Center.

Secure Web (HTTPS)

Secure Web (HTTPS) through Proxy
Enter the Hostname/IP Address and port number for Proxy

Hostname/IP Address—Enter the IP address or the hostname of the
proxy server to send the Call Home messages through an indirect network
connection.

Port—Enter the port number on which the proxy server is enabled.

Email

Note

You must have configured the SMTP for Email to be sent successfully.

To
send a copy of the Call Home messages to multiple email recipients, enter the
email addresses separated with a comma. You can enter up to a maximum of 1024
characters.

Click
Continue to proceed, or select
Back to return to the previous menu. If you click
Continue, a message appears as
To help improve the Cisco
Unified Communications Manger experience, click Confirm to allow Cisco Systems
to securely receive usage statistics from the server. This information will be
used by Cisco to help understand how customers are using our product and
ultimately drive product direction. If you prefer not to participate, you may
choose to opt-out.

Click
Confirm to proceeds with normal installation or
select
Back to return to the Smart Call Home Enable Page.

Select
Remind me
Later to configure Smart Call Home to configure the Smart Call Home service
after installation, using Cisco Unified Serviceability pages.

A reminder
message appears in Cisco Unified CM Administration.

Smart Call Home is not configured. To configure Smart Call Home
or disable the reminder, please go to Cisco Unified Serviceability > Call
Home.

Select
Disable All
Call Home on System Start to disable the Smart Call Home service. However,
you can activate the Smart Call Home service after installation using Cisco
Unified Serviceability pages.

Note

You can
reconfigure the service in Cisco Unified Serviceability page after
installation. For more information, see the
Cisco
Unified Serviceability Administration Guide.

Step 7

Choose
OK. The Application User Configuration window
appears.

Step 8

Enter the
Application User name and password from and confirm the password by entering it
again.

Step 9

Choose
OK. The Platform Configuration Confirmation window
appears.

Step 10

To continue with
the installation, choose
OK; or to modify the platform configuration, choose
Back.

The system
installs and configures the software. The server reboots.

When the
installation process completes, you are prompted to log in by using the
Administrator account and password.

Call Secure Status
Policy

For
Unified Communications Manager, Release 10.0(1) and
later, Call Secure Status Policy controls display of secure status icon on
phones. The following are the policy options:

All media
except BFCP and iX application streams must be encrypted.
This is the
default value. The security status of the call is not dependent on the
encryption status of BFCP and iX application streams.

All media
except iX application streams must be encrypted.
The security
status of the call is not dependent on the encryption status iX application
streams.

All media
except BFCP application streams must be encrypted.
The security
status of the call is not dependent on the encryption status BFCP.

All media in a
session must be encrypted.
The security
status of the call is dependent on the encryption status of all the media
streams in an established session.

Only Audio
must be encrypted.
The security
status of the call is dependent on the encryption of the audio stream.

Note

Changes to
the policy affects the display of the secure icon and the playing of a secure
tone on the phone.

A new service
parameter, Secure Call Icon Display Policy, manages the call secure status
policy. It replaces the following service parameters:

Cisco Unified
Communications Manager Administration Considerations

Bulk
Administration Considerations

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Caller-Specific
Music On Hold

Cisco
Unified Communications Manager can play a different MOH audio source
for SIP calls that a phone receives over the SIP trunk, which are then put on
hold, depending on the MOH stream IDs that are added to the SIP header for the
call. An external application, such as the Cisco Unified Customer Voice Portal
(CVP) contact center solution, adds the MOH stream IDs for user and network
hold to the SIP header, and then relays that to
Cisco Unified Communications Manager over the SIP
trunk.

Cisco Unified
Communications Manager Administration Considerations

If the incoming
SIP call contains MOH audio source information in the SIP header, then
Cisco Unified Communications Manager initiates the
following actions:

The MOH audio
source is played for the caller when the SIP call is placed on user hold.

The MOH audio
source is played for the caller when the SIP call is placed on network hold.

The MOH audio
source is played for the caller if the call is transferred to another endpoint
on the same cluster and subsequently placed on user or network hold.

When a call is
sent on a SIP trunk to another cluster, the MOH audio source information is
sent along with the call.

When a call is
sent on a SIP trunk to another cluster in an SME scenario, the MOH audio source
information is sent along with the call.

When a call
is transferred to another cluster over a SIP trunk, the MOH audio source
information is sent along with the call.

When a call is
either forwarded or redirected to another cluster over a SIP trunk, the MOH
audio source information is sent along with the call.

Limitations

If the user
and network MOH audio source identifiers are not provisioned, or if one or both
values are invalid, the caller-specific MOH information in the SIP header is
ignored. The call reverts to tone on hold and an invalid MOH audio source alarm
is raised.

When both the
user and network MOH audio source identifiers are present in the header, any
invalid value is replaced by the default value (0).

If both values
are 0, or the only value is 0, the header in the incoming INVITE is ignored.

If only one
MOH audio source identifier is provided in the SIP header, including if a comma
appears before or after the MOH audio source identifier value, the same MOH ID
is used for both user and network MOH. The SIP trunk populates both the user
and the network MOH audio source identifiers in the SIP header so that Call
Control always receive both values.

If there are
more than two MOH audio source identifier values separated by a comma in the
header, then the first two values are used. Subsequent values are ignored.

The
original incoming caller to the call center cannot change during the course of
the entire call.

The music
on hold information is only shared across SIP trunks.

Caller-specific MOH is not supported when calls are received or
transferred over QSIG tunneling-enabled SIP trunks.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Additional
Information

Three new alarms
are added for caller-specific Music On Hold:

OutOfRangeMohAudioSource

UnableToOpenMohAudioSource

UnprovisionedMohAudioSource

Certificate
Revocation Check

For
Cisco Unified Communications Manager, Release 10.0(1),
you can configure certification revocation check to be performed on a periodic
basis. Certificate revocation check is performed on all certificates and trust
chains associated with a established long session. The check is performed for
the following sessions:

The check
terminates established sessions when the certificate or trust chain status is
revoked, not trusted or expired.

The enterprise
parameter
Certificate Revocation and Expiry allows you to
control the certificate validation checks. The certificate service frequently
checks for long sessions between
Cisco
Unified Communications Manager and other services. The certificate
expiry for the long sessions is not verified when the
Certificate Revocation and Expiry parameter value is
disabled.

Choosing
Enable
Revocationon the Operating System Administration of
Cisco
Unified Communications Manager activates the certificate revocation
service for LDAP and IPsec connections.
Check
Every value determines the frequency of the check. If the
Enable
Revocation check box is unchecked then the revocation check for the
certificate is not performed.

The following
fields are added to the
Certificate Revocation window (Security > Certificate
Revocation) in
Cisco
Unified Operating System Administration interface:

Cisco Unified
Communications Manager Administration Considerations

Bulk
Administration Considerations

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Cisco AXL Web
Service Enabled by Default

With Release
10.0(1), Cisco AXL Web Service is now enabled by default on all cluster nodes
following installation. Cisco recommends that you always leave the service
activated on the publisher node to ensure that you are able to configure
products that are dependent on AXL, such as Unified Provisioning Manager.

Based on your
needs, you can start or stop the service on specific subscriber nodes in Cisco
Unified Serviceability under Feature Services.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

Following
installation, Cisco AXL Web Service is now enabled by default on all cluster
nodes.

The Media
Termination Point (MTP) device, software conference bridge, annunciator, and
unicast Music On Hold provided by the Cisco IP Voice Media Streaming
Application service support both IPv4 and IPv6 audio media connections. The MTP
device, software conference bridge, annunciator, and unicast Music On Hold are
configured automatically in dual mode when the platform is configured for IPv6
and the IPv6 enterprise parameter is enabled. If the platform is not configured
for IPv6, the MTP device, software conference bridge, annunciator, and unicast
Music On Hold are configured automatically in IPv4 only mode.

The MTP device,
software conference bridge, annunciator and Music On Hold support only IPv4 for
the TCP control channel. The annunciator, Music On Hold, and MTP in
pass-through mode support secure media SRTP connections to both IPv4 and IPv6
addresses.

Note

Multicast Music
On Hold supports only IPv4.

Note

Both Cisco IOS
Enhanced MTP and the software MTP provided by the Cisco IP Voice Media
Streaming Application support IPv4 to IPv6 translation. Both Cisco IOS Enhanced
MTP and the software MTP provided by the Cisco IP Voice Media Streaming
Application support media interoperation between IPv4 and IPv6 networks and
operate in dual mode. For information on how
Cisco Unified Communications Manager (Unified Communications Manager) inserts MTPs into
calls that require IPv4 to IPv6 translation, see
Cisco
Unified Communications Manager Features and Services Guide. For
information on how to configure your Cisco IOS MTP so that the MTP can support
IP translation, see
Implementing VoIP for IPv6.

Conferences

Unified Communications Manager supports dual mode for
the software conference bridge provided by the Cisco IP Voice Media Streaming
Application. During a conference, if an endpoint supports IPv4 only, IPv4 media
is negotiated between the endpoint and the conference bridge. Whereas, if the
endpoint supports IPv6 only, IPv6 media is negotiated between the endpoint and
the conference bridge. If dual mode is supported by the SCCP endpoint, the
media preference configured in the enterprise parameter (IPv4 or IPv6) is
negotiated between the endpoint and the conference bridge. If dual mode with
ANAT is supported by the SIP device, the ANAT address preference advertised by
the SIP device is negotiated between the SIP device and the conference bridge.
For conferences using the software conference bridge,
Unified Communications Manager does not insert an MTP
for IPv4 to IPv6 translation because the software conference bridge supports
dual mode conferences.

If an MTP is
inserted in a conference, for it to support security you must configure the MTP
in pass-through mode, which means that the MTP does not transform the media
payload during the call. When you configure an MTP in pass-through mode, the
MTP receives the encrypted packet on one call leg and sends out the same packet
on a different leg of the call. For secure conferences with secure conference
bridges that do not support dual mode and encrypted devices with an IP
Addressing Mode of IPv6 Only,
Unified Communications Manager inserts an MTP into the
conference to translate IPv4 to IPv6 (and vice versa). If you configure the MTP
for pass-through mode, the encrypted IPv6 phones communicate with the
conference bridge using SRTP. If you do not configure the MTP for pass-through
mode, the media gets downgraded to RTP.

Music On
Hold

The Cisco IP Voice
Media Streaming Application, which is a component of Music On Hold, supports
both IPv4 and IPv6 audio media connections for unicast Music On Hold. Multicast
Music On Hold supports IPv4 only. So, devices with an IP Addressing Mode of
IPv6 Only cannot support multicast Music On Hold. Under these circumstances,
Unified Communications Manager plays a tone, instead
of music, when the phone is on hold. However, devices with an IP Addressing
Mode of IPv6 only can stream unicast Music On Hold without
Unified Communications Manager inserting an MTP for
IPv4 to IPv6 conversion.

CDR/CAR
Considerations

IP Phones
Considerations

RTMT
Considerations

Security
Considerations

Serviceability
Considerations

CLI Changes

In order to support upgrade and migration, the Refresh Upgrade framework is being used for export the data from the old system and import the data to the new virtual system. In a system where export operation is not supported by default, the user needs to install a Cisco Options Package (COP) file which will install required files and make changes to the command-line interface (CLI).

Three CLI commands are added when the COP file is installed:

utils system dataexport initiate

utils system dataexport cancel

utils system dataexport status

The utils system dataexport initiate command can be used for initiating the export operation. The user is prompted for the SFTP server details. After the export starts, the command is executed in the back end and the CLI returns.

Cisco Prime Collaboration Deployment simplifies the hostname and IP address change procedures. Hostname change refers to the changing of a server network hostname identity, an IP address, or both. Historically, changing a server hostname involved a series of steps that required the use of the CLI and web graphical user interface (GUI). It also required the user to reboot single servers and whole clusters after the event.

All the existing steps in the procedure "Changing the IP Address and Hostname for Cisco Unified Communications Manager" are automated into a single step using the CLI or GUI. The single-step includes a readiness and post-task list for user verification.

Also new as part of Cisco Prime Collaboration Deployment is the enhancement of the "set network hostname" CLI command. The interface now prompts the user for an IP, mask, and gateway value.

Procedure Changes

Procedures for "Changing the IP Address and Hostname for Cisco Unified Communications Manager" are affected (see CLI changes above). Cisco Prime Collaboration Deployment is used to simplify and automate IP address and hostname changes across the cluster as part of a software upgrade, server migration, or re-numbering on a Release 10.0 (or higher) system.

New procedures have been created for migrating data from an existing Cisco Unified Communications Manager node to a new machine. This operation was previously done using the Bridge Upgrade procedure. These steps can be done automatically, using the Cisco Prime Collaboration Deployment application (a new document, Cisco Prime Collaboration Deployment Administration Guide, will be delivered for Release 10.0(1)).

Important: Cisco Prime License Manager is not installable as a standalone
option from the Unified Communications operating system ISO. Co-resident
installation from the Unified Communications operating system ISO is still an
option.

Note

You have the
ability to define how to manage licensing of your enterprise. You can have one
Cisco Prime License Manager for the entire enterprise,
or you can have several
Cisco Prime License Managers and divide the enterprise
in a manner that best suits your needs.

Cisco
Prime License Manager runs on a virtual machine or may co-reside on a
product's virtual machine if that product supports co-resident deployment.

IP Phones
Considerations

RTMT
Considerations

Security
Considerations

Serviceability
Considerations

Cisco TelePresence MCU Settings

Cisco TelePresence MCU refers to a set of hardware
conference bridges for
Cisco Unified Communications Manager.

The Cisco TelePresence MCU is a high-definition (HD) multipoint video conferencing bridge. It delivers up to 1080p at 30 frames per second, full continuous presence for all conferences, full trans-coding, and is ideal for mixed HD endpoint environments.

The Cisco TelePresence MCU supports SIP as the signaling call control protocol. It has a built in Web Server that allows for complete configuration, control and monitoring of the system and conferences. The Cisco TelePresence MCU provides XML management API over HTTP.

See the
Cisco Unified Communications Manager System Guide for more
information about conference bridges.

The following table describes the Cisco TelePresence MCU
configuration settings.

The following table describes the Cisco TelePresence MCU configuration settings.

Table 4 Cisco TelePresence MCU Configuration Settings

Field

Description

Conference Bridge Name

Enter a name for your conference bridge

Destination Address

Enter the IP Address of the Cisco TelePresence MCU conference
bridge

Description

Enter a description for your conference bridge

Device Pool

Choose a device pool or choose Default.

Common Device Configuration

Choose the common device configuration to assign to the
conference bridge. The common device configuration includes attributes, such as
MOH audio source, that support features and services for phone users.

Device configurations that are configured in the Common Device
Configuration window display in the drop-down list.

Location

Use location to implement call admission control (CAC) in a
centralized call-processing system. CAC enables you to regulate audio quality
and video availability by limiting the amount of bandwidth that is available
for audio and video calls over links between locations. The location specifies
the total bandwidth that is available for calls to and from this location.

From the drop-down list box, choose the appropriate location
for this conference bridge.

A location setting of Hub_None means that the locations
feature does not keep track of the bandwidth that this conference bridge
consumes. A location setting of Phantom specifies a location that enables
successful CAC across intercluster trunks that use H.323 protocol or SIP.

To configure a new location, use the
System > Location
menu option.

For an explanation of location-based CAC across intercluster
trunks, see the
Cisco Unified Communications Manager System Guide.

Use Trusted Relay Point

From the drop-down list box, enable or disable whether Cisco
Unified CM inserts a trusted relay point (TRP) device with this media endpoint.

Default—If you
choose this value, the device uses the Use Trusted Relay Point setting from the
common device configuration with which this device associates.

Off—Choose this
value to disable the use of a TRP with this device. This setting overrides the
Use Trusted Relay Point setting in the common device configuration with which
this device associates.

On—Choose this
value to enable the use of a TRP with this device. This setting overrides the
Use Trusted Relay Point setting in the common device configuration with which
this device associates.

A Trusted Relay Point (TRP) device designates an MTP or
transcoder device that is labeled as Trusted Relay Point.

Cisco Unified CM places the TRP closest to the associated
endpoint device if more than one resource is needed for the endpoint (for
example, a transcoder or RSVPAgent).

If both TRP and MTP are required for the endpoint, TRP gets
used as the required MTP. See the
Cisco Unified Communications Manager System Guide for details of call behavior.

If both TRP and RSVPAgent are needed for the endpoint, Cisco
Unified CM first tries to find an RSVPAgent that can also be used as a TRP.

If both TRP and transcoder are needed for the endpoint, Cisco
Unified CM first tries to find a transcoder that is also designated as a TRP.

See the
Cisco Unified Communications Manager System Guide for a complete discussion of network virtualization and
trusted relay points.

SIP Interface Info

MCU Conference Bridge SIP Port

This is the SIP listening port of the Cisco TelePresence MCU
Conference Bridge. The default value is 5060.

From the drop-down list box, choose the security profile to apply to the SIP trunk.

You must apply a security profile to all SIP trunks. Cisco Unified Communications Manager provides a default nonsecure SIP trunk security profile for autoregistration. To enable security features for a SIP trunk, configure a new security profile and apply it to the SIP trunk. If the trunk does not support security, choose a nonsecure profile.

Check this check box to enable tracing within the script or uncheck this check box to disable tracing. When checked, the trace.output API provided to the Lua scripter produces an SDI trace.

Note

Cisco recommends that you enable tracing only while debugging a script. Tracing has an impact on performance and should not be enabled under normal operating conditions.

Parameter Name/Parameter Value

Optionally, enter parameter names and parameter values. Valid values include all characters except equal signs (=), semicolons (;), and nonprintable characters, such as tabs. You can enter a parameter name with no value.

You must choose a script from the Normalization Script drop-down list box before you can enter parameter names and values.

To add another parameter line, click the + (Plus) button. To delete a parameter line, click the – (Minus) button.

HTTP Interface Info

Username

Enter the Cisco TelePresence MCU administrator username.

Password

Enter the Cisco TelePresence MCU administrator password.

Confirm Password

Enter the Cisco TelePresence MCU administrator password

HTTP Port

Enter the Cisco TelePresence MCU HTTP port. The default port
is 80.

Use HTTPS

Check this check box if you want to create a secure HTTPS connection between Cisco Unified Communications Manager and Cisco TelePresence MCU. The default HTTPS port is 443.

For information on how to create a TLS connection between Cisco Unified Communications Manager and Cisco TelePresence MCU, see the Set up a TLS connection with Cisco TelePresence MCU section.

Note

The HTTP configuration must match what is configured on the Cisco
TelePresence MCU. This information allows
Cisco Unified Communications Manager to invoke the remote management API on
the Cisco TelePresence MCU.

Cisco TelePresence Conductor Settings

Cisco TelePresence
Conductor provides intelligent conference administrative controls and is scalable, supporting device clustering for load balancing across
MCUs and multiple device availability. Administrators can implement the Cisco
TelePresence Conductor as either an appliance or a virtualized
application on VMware with support for Cisco Unified Computing
System (Cisco UCS) platforms or third-party-based
platforms. Multiway conferencing, that allows for dynamic two-way to three-way conferencing, is also supported.

Cisco TelePresence Conductor supports both ad hoc and meet-me voice
and video conferencing. Cisco TelePresence Conductor
dynamically selects the most appropriate Cisco TelePresence
resource for each new conference. Ad hoc, "MeetMe" and scheduled voice and video conferences can dynamically grow and exceed the
capacity of individual MCUs. One Cisco
TelePresence Conductor appliance or Cisco TelePresence Conductor
cluster has a system capacity
of 30 MCUs or 2400 MCU ports.
Up to three Cisco TelePresence Conductor appliances or
virtualized applications may be clustered to provide
greater resilience.

Cisco
TelePresence Conductor also provides the XML management API over HTTP, and has a built-in Web Server for complete
configuration, control and monitoring of the system and conferences. For more information, see the Cisco TelePresence Conductor Administrator Guide and the Cisco Unified Communications Manager System Guide.

Note

If you are using encryption with Cisco TelePresence Conductor, select cisco-telepresence-conductor-interop as the
default normalization script.

The following table describes the Cisco TelePresence Conductor
configuration settings.

Table 5 Cisco TelePresence Conductor Configuration Settings

Field

Description

Conference Bridge Name

Enter a name for your conference bridge.

Description

Enter a description for your conference bridge.

Conference Bridge Prefix

The Conference Bridge Prefix is used only for centralized deployments when the conference resources are deployed across a Small Medium Enterprise (SME) and the HTTP and SIP signaling are intended for different destinations.

Do not set this parameter unless your video conference device supports this function. See the documentation that came with your conference bridge device for details.

SIP Trunk

Select a SIP trunk to use for this conference bridge from a list of existing SIP trunks.

HTTP Interface Info

Override SIP Trunk Destination

Check this check box to override the SIP trunk destination. Use this feature if the HTTP and SIP signaling are intended for different destination IP addresses, for example, when the device is used in a centralized deployment. Click the "+" and "-" buttons to add or remove IP addresses and hostnames.

Do not set this parameter unless your video conference device supports this function. See the documentation that came with your conference bridge device for details.

Hostname/IP Address

Enter one or more hostnames or IP addresses for the HTTP signaling destination if you have selected to override the SIP trunk destination.

Username

Enter the Cisco TelePresence Conductor administrator username.

Password

Enter the Cisco TelePresence Conductor administrator password.

Confirm Password

Enter the Cisco TelePresence Conductor administrator password

Use HTTPS

Check this check box if you want to create a secure HTTPS connection between Cisco Unified Communications Manager and Cisco TelePresence Conductor. The default HTTPS port is 443.

For improved performance, use the default standard SIP profile for TelePresence conferencing that has the Options ping configured.

Assign the encryption interworking script
to SIP trunks that are used for Cisco TelePresence Conductor if encryption is used.

See topics related to setting up trunks for more details about SIP trunk configuration.

Limitations

Media Termination Point (MTP)
Required: Cisco Unified Communications Manager ignores this configuration for all ad hoc
conference calls even if this is selected on the SIP trunk.

Early Offer Support for Voice and Video calls: Cisco Unified Communications Manager ignores
this configuration for all ad hoc conference calls even if this is
selected on the SIP profile that is associated with the SIP trunk that is linked to the conferencing resource server.

SIP
Rel1xx Option: Cisco Unified Communications Manager ignores this configuration for ad hoc
conference calls even if this is enabled on the SIP profile
associated with the SIP trunk that is linked to the conferencing resource server.

RSVP over SIP: Cisco Unified Communications Manager ignores this configuration for all ad hoc conference calls
if this is enabled for E2E. If this is configured for local
RSVP, the configuration will be effective.

Procedure changes

Set Up TelePresence Video Conference Bridge

Use Cisco Unified Communications Manager Administration to add and configure a video conference bridge device. Each video conference bridge device must be assigned to a SIP trunk when you configure the video conference device for the node.

Before You Begin

Set up a SIP trunk before you proceed. See topics related to trunk setup and SIP trunk setup for video conference bridge devices for details.

Cisco Unified
Communications Manager Administration Considerations

Bulk
Administration Considerations

CDR/CAR
Considerations

IP Phones
Considerations

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Commercial Cost
Avoidance

Cisco Unified
Communications Manager Administration Considerations

If a caller within
a
Cisco Unified Communications Manager (Unified Communications Manager) network calls a called
party on an external number,
Unified Communications
Manager checks if an internal number exists for the called party in
the LDAP database. If an internal number exists, the call is routed to the
internal number. If the internal number is not found in the LDAP database, the
call is routed to the original (external) number.

To route the calls
to the internal numbers, you must configure directory number alias for both the
lookup and the sync servers. You must configure the LDAP server for Directory
Number Alias Sync (sync server) to synchronize users from
Unified Communications
Manager database to the sync server. You must configure the LDAP
server for Directory Number Alias Lookup (lookup server) to route the
commercial calls to an alternate number.

In
Cisco Unified
Communications Manager Administration, use the submenus under the
Advanced
Features > Directory Number Alias Lookup/Sync
menu path to configure directory number alias lookup and sync servers. .

The following
table describes the Directory Number Alias Lookup/Sync settings.

Table 6 Directory
Number Alias Lookup/Sync Settings

Field

Description

LDAP Directory
Information

LDAP
Configuration Name

Enter a
unique name (up to 40 characters) for the LDAP directory.

LDAP
Manager Distinguished Name

Enter
the user ID (up to 128 characters) of the LDAP Manager, who is an
administrative user that has access rights to the LDAP directory in question.

LDAP
Password

Enter a
password (up to 128 characters) for the LDAP Manager.

Confirm
Password

Reenter
the password that you provided in the LDAP Password field.

LDAP
User Search Base

Enter
the location (up to 256 characters) where all LDAP users exist. This location
acts as a container or a directory. This information varies depending on
customer setup.

LDAP
Directory Server Usage

Specify
if the LDAP directory server should be used as:

Directory Number Alias Sync and Lookup

Directory Number Alias Sync Only

Directory Number Alias Lookup Only

By
default, the Directory Number Alias Sync and Lookup option is selected. If you
choose the
Directory Number Alias Sync and Lookup option, you cannot
add another sync or lookup server.

Directory Number Alias
Server Configuration

Keepalive Search User Distinguished Name

Enter
the user ID (up to 128 characters) of the administrative user for which you
need to perform the keepalive search.

Keepalive Time Interval in Minutes

Specify
the time interval at which keepalive messages should be sent to lookup/sync
servers to check if those servers are active or not.

For
example, if you specify the keepalive time interval as 10 minutes and select
the LDAP directory server as
DN
Alias Lookup only, keepalive messages will be sent every 10 minutes to
all the lookup servers that are configured.

Enable
Caching of Records for Directory Number Alias Lookup

Check
this check box to enable caching of records for directory number alias lookup.
If you check this check box, you can specify Record Cache Size for Directory
Number Lookup Alias and Record Cache Age for Directory Number Alias Lookup in
Hours.

Note

If you
specify the LDAP directory server as a sync server, the system disables this
check box.

This
field is enabled only if the Lookup server or both (Lookup and Sync) the
servers are used as LDAP directory servers. If the Sync server is used as LDAP
directory server, this field is disabled.

Record
Cache Size for Directory Number Alias Lookup

Specify
the number of records that should be cached. You can specify any number within
a range of 3000-10000.

Note

This
field is enabled only if 'Enable Caching of Records for Directory Number Alias
Lookup' check box is checked.

Record
Cache Age for Directory Number Alias Lookup in Hours

Specify
the time for which the records should be held in the record cache.

Note

This
field is enabled only if 'Enable Caching of Records for Directory Number Alias
Lookup' check box is checked.

LDAP Server
Information

Host
Name or IP Address for Server

Enter
the hostname or IP address of the server where the data for this LDAP directory
resides.

Port

Enter
the port number on which the LDAP routing database receives the LDAP requests.

How your
LDAP routing database is configured determines which port number to enter in
this field. For example, before you configure the LDAP Port field, determine
whether your LDAP server acts as a Global Catalog server and whether your
configuration requires LDAP over SSL. Consider entering one of the following
port numbers:

LDAP
Port when LDAP server is not a Global Catalog server:

389:
When SSL is not required. (This port number specifies the default that displays
in the LDAP Port field.)

636:
When SSL is required. (If you enter this port number, make sure that you check
the Use SSL check box.)

LDAP
Port when LDAP server Is a Global Catalog server:

3268: When SSL is not required.

3269: When SSL is required. (If you enter this port number, make
sure that you check the Use SSL check box.)

Tip

Your configuration may require that you enter a different port
number than the options that are listed in the preceding bullets. Before you
configure the LDAP Port field, contact the administrator of your directory
server to determine the correct port number to enter.

Add
Another Redundant LDAP Server

Click
this button to add a redundant LDAP server.

Note

To enable
routing the commercial calls to the internal numbers of the called parties,
ensure that Cisco Directory Number Alias Lookup Service is activated. To
synchronize users from the
Unified Communications
Manager database to the LDAP server for Directory Number Alias Sync
server, ensure that Cisco Directory Number Alias Sync Service is activated.

Note

You can
configure the primary and secondary lookup and sync servers to support
failover. If a primary server goes down and if the secondary server is
configured, lookup/sync services automatically connect to the secondary server.
The failover is supported for both lookup and sync services. When the primary
server is restored, the network administrator must restart the lookup/sync
service so that the services can connect back to the primary server.

Note

A commercial
call is routed to an internal number only if Confidential Access Level (CAL)
resolution succeeds on that call. If the CAL resolution fails, the call is
redirected to the original destination.

Call Control
Agent Profile Configuration—This menu path has been added to enable you to
configure the call control agent profile settings.

The following
table describes the Call Control Agent Profile settings.

Table 7 Call Control
Agent Profile Settings

Field

Description

Call Control Agent
Profile Configuration

Call
Control Agent Profile ID

Enter
the Call Control Agent Profile ID.

Primary
Softswitch ID

Enter
the primary softswitch ID.

Secondary Softswitch ID

Enter
the secondary softswitch ID.

Object
Class

Enter
the object class name to be synchronized to the external directory server.

Subscriber Type

Enter
the subscriber type.

SIP
Alias Suffix

Enter
the SIP alias suffix. The E.164 number that you specify for the directory
number is appended to this suffix.

SIP User
Name Suffix

Enter
the SIP user name suffix.

A new field
Call
Control Agent Profile has been added to the Directory Number
Configuration window (Call
Routing > Directory Number). You can select the
Call Control Agent Profile to associate to the directory number.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

The following
services have been added in the Control Center - Feature Services window
(Tools > Control Center - Feature
Services):

Select a Call Control Agent Profile from the drop-down list box to create a new Call Control Agent Profile.

What to Do Next

Configure the LDAP server for Directory Number Alias Sync (sync server) if you need to synchronize directory numbers from the Unified Communications Manager database to the sync server.

Common Cluster
Topology

IM and Presence
Service administration functions have been integrated into the
Cisco Unified Communications Manager Administration.
Common cluster topology is one of two main areas of this integration, the other
being common user management which is documented separately.

Components of the
common topology integration for this release include:

Add/edit IM
and Presence Service nodes

Add/edit
presence redundancy groups and high availability

IM and
Presence Service node information and status

IM and
Presence Service user assignments and status

Select cluster
node type (Unified CM or IM and Presence)

Unified CM
Cluster overview report

Cisco Unified
Communications Manager Administration Considerations

The following new
functions are accessible using
Cisco Unified Communications Manager Administration.
For more information and detailed procedures, see the
Cisco Unified Communications Manager Administration
Guide and the
Cisco
Unified Communications Manager
Features and Services Guide.

IM
and Presence Service node setup and status

Note

To install
the
IM and Presence
Service node, see
Installing Cisco Unified Communications Manager.

High
Availability and presence redundancy group setup

Manual
failover, fallback, and recovery

Balancing user
and server assignments

End user setup
for
IM and Presence
Service

Bulk Administration
Considerations

A new field to
assign an end user to an
IM and
Presence Service node is added to the BAT update users and the BAT
user template field descriptions tables.

Assigned
Presence Server

Assign
the end user to an
IM and Presence
Service node that is installed in the cluster if the system is
non-balanced. The server you specify using the Bulk Administration Tool must be
part of a Presence Redundancy Group.

For
clusters that have the user assignment mode for the IM and Presence Service
node set to balanced or active-standby, user assignments that are made using
the Bulk Administration Tool override the automatic user assignments.

CDR/CAR
Considerations

The IM and
Presence Cluster Overview report information is included in the Unified CM
Cluster Overview report. The IM and Presence Cluster Overview report is no
longer available as a separate report.

If you are
prompted to re-login when you select an
IM and Presence Service report, re-enter your
Cisco Unified Communications Manager Administration
Administration login credentials

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

IM and Presence Service
Considerations

You can no longer
create presence redundancy groups or add users to presence redundancy groups on
IM and
Presence Service using the Cisco Unified CM IM and Presence
Administration GUI. Presence Topology, which was renamed from Cluster Topology,
now provides a read-only view of topology settings.

utils ha failover

This command initiates a manual failover for a specified node, where the Cisco Server Recovery Manager stops the critical services on the failed node and moves all users to the backup node.

For IM and Presence nodes, the backup node must be another IM and Presence server. Two servers must be assigned to the same presence redundancy group before you specify the backup server. The back-up server you specify is the other server that is assigned to the presence redundancy group.

utils ha recover

This command initiates a manual recovery of the presence redundancy group (when nodes are in a Failed state), where IM and Presence restarts the Cisco Server Recovery Manager service in that presence redundancy group.

utils ha recover presence redundancy group name

Syntax Description

Parameters

Description

presence redundancy group name

Specifies the presence redundancy group on which to monitor HA status. If no presence redundancy group name is provided, all cluster information is provided.

Status Example Failed

Common
Serviceability

Cisco Unified
Communications Manager Administration Considerations

Cisco
Unified Communications Manager (Unified Communications
Manager), Release 10.0(1) provides a common serviceability for
Unified Communications Manager and IM and Presence
nodes in a mixed cluster setup. The following modules have been integrated to
give you a unified administration and reporting experience:

Cisco Unified
Serviceability

Cisco Unified
Real-Time Monitoring Tool (RTMT)

Alert Manager
and Controller Service (AMC)

No changes.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

The following
changes have been made in the RTMT interface:

A new IM and
Presence drawer has been added, which displays information on only IM and
Presence nodes.

The
CallManager drawer has been renamed as Voice/Video. It displays information on
only voice or video nodes.

System Drawer
- The System Drawer has the following changes:

You can
view the summary for all the nodes in the extended cluster (Voice/Video or IM
and Presence) on the System Summary screen.

You can
view information for all the nodes in the extended cluster (Voice/Video or IM
and Presence) for CPU and Memory, Process, and Disk Usage.

You can
view performance information from all nodes in the extended cluster
(Voice/Video or IM and Presence). The applicable counters will be displayed
based on the type of node that you select.

You can
view alerts for the system for all nodes in the extended cluster.

You can
view real-time trace and monitor use events for all the nodes in the extended
cluster.

You can
display audit logs from all nodes in the extended cluster.

Security
Considerations

No changes.

Serviceability
Considerations

The following
changes have been made in the Cisco Unified Serviceability interface:

Common User
Management

The following user
management enhancements are added for
IM and Presence Service on
Cisco
Unified Communications Manager (Unified Communications
Manager):

End user
meeting and calendar information can be included in IM and Presence Service.

Presence
Viewer allows users to view the availability of their watchers and contacts, as
well as access information about their current presence server assignment.

IM
and Presence Service roles are added to
Unified Communications
Manager.

Cisco Unified
Communications Manager Administration Considerations

Calendar and Meeting
Information Inclusion

You can enable the
inclusion of end user meeting and calendar information in
IM and Presence Service from either the End User
Configuration or the Feature Group Template Configuration windows in
Unified Communications
Manager Administration. The following conditions must be met to
enable this feature:

The end user
must be on the home cluster and have
IM and Presence
Service enabled.

An Exchange
Presence Gateway must be configured on the
Cisco
Unified Communications ManagerIM and Presence
Service server.

IM
and Presence Service Roles

The following
IM and Presence
Service roles have been added to
Unified Communications
Manager

Table 8 IM and
Presence Service Roles

Standard
Role

Supported
Application(s)

Privileges/Resources for the Role

Associated Standard User Group(s)

Standard
CCMADMIN Administration

Cisco
Call Manager IM and Presence Administration

Allows
an administrator access to all aspects of the CCMAdmin system

Standard
CCMADMIN Read Only

Cisco
Call Manager IM and Presence Administration

Allows
read access to all CCMAdmin resources

Standard
CUReporting

Cisco
Call Manager IM and Presence Reporting

Allows
application users to generate reports from various sources

Bulk
Administration Considerations

You can enable the
inclusion of end user meeting and calendar information in
IM and Presence
Service using the Bulk Administration Tool in the BAT user template
and in the User update settings. The user must be on the home cluster and have
IM and Presence
Service enabled. Also ensure that an Exchange Presence Gateway is
configured on the
Cisco
Unified Communications Manager IM and Presence
Service server.

CDR/CAR
Considerations

Tip

When generating
a new report for IM and Presence Service, reenter your
Cisco Unified
Communications Manager Administration login credentials if you are
prompted to log in again when you select an
IM and
Presence Service report to view.

The following
reports have been added for
IM and
Presence Service:

Table 9 IM and
Presence Service Reports

Report

Description

Presence
Configuration Report

Provides
configuration information about
IM
and Presence Service users.

Users that are synced from
Cisco Unified Communications Manager

Users
that are enabled for
IM and Presence Service

Users
that are enabled for Microsoft remote call control

Users
that are enabled for calendaring information in
IM and Presence Service

IP Phones
Considerations

RTMT
Considerations

Security
Considerations

Serviceability
Considerations

Presence Viewer for End Users

Use the Presence Viewer to view the availability status of a user in
IM and Presence Service, and to view the list of contacts and watchers that are configured for that user.

Access the Presence Viewer from an end-user configuration record using Cisco Unified Communications Manager Administration when IM and Presence Service is enabled for that user. For more information, see topics related to enablingIM and Presence Service for a user.

The user must be assigned to
an IM and Presence Service node for valid presence information to be available. The AXL, Presence Engine, and Proxy Service must all be running on
the IM and Presence Service node for this feature to be functional.

The following table lists the fields that are displayed on the Presence Viewer for the selected end user in Cisco Unified Communications Manager Administration.

Table 10 End User Presence Viewer Fields

Configuration/Availability Information

User Status

Identifies the availability state of the user,
including:

Available

Away

Do Not Disturb

Unavailable

Custom

User ID

Identifies the selected user ID.
A user photo is displayed if one is available for that user.

You can click Submit to choose a different User ID.

View From Perspective of

Specifies a user to see the availability status from the perspective of the user. This allows you to determine how the availability status of a specified user appears to another user, known as a watcher. This functionality is useful in debugging scenarios, for example, where a user has configured privacy policies.

A maximum of 128 characters is allowed.

Contacts

Displays the number of contacts in the contact list for this user.

Click the arrow beside the Contacts heading in the Contacts and Watchers list area to view the availability status of
a specific user contact. Click the arrow beside the group name to expand the list of contacts within that group.

Contacts that are not part of a group (groupless contacts) display below the
contact group list. A contact may belong to multiple groups, but will only count once against the
contact list size for that user.

A warning message appears if the maximum number of contacts configured for end users is exceeded. For more information about IM and Presence Service configuration and the maximum contacts setting, see the IM and Presence Administration Online Help.

Watchers

Displays a list of users, known as watchers, who
have subscribed to see the availability status of this user in their contact list.

Click the arrow beside the Watchers heading in the Contacts and Watchers list area to view the availability status of
a specific watcher. Click the arrow beside the group name to expand the list of watchers within that group.

A watcher may belong to
multiple groups but will only count once against the watcher list size for that
user.

A warning message appears if the maximum number of watchers configured for end users is exceeded. For more information about IM and Presence Service configuration and the maximum watchers setting, see the IM and Presence Administration Online Help.

Presence Server Assignment

Identifies the IM and Presence Service server
to which the user is assigned. Hyperlinks allow you to go directly
to the server configuration page for details.

Enable accessible presence icons

Select this check box to enable presence accessibility icons for this end user.

Submit

Select to run the Presence Viewer.

The user must be assigned to
an IM and Presence node for valid presence information to be available.
The AXL, Presence Engine and Proxy Service must all be running on
the IM and Presence server for this action to be functional.

Procedure changes

Display Presence Viewer for End Users

Use Cisco Unified Communications Manager Administration to display the Presence Viewer for an end user.

Before You Begin

The end user must be on the home cluster and have IM and Presence enabled.

Ensure that an Exchange Presence Gateway is configured on the Cisco Unified Communications Manager IM and Presence Service server.

Procedure

Step 1

Select User Management > End User to find the end user.

The End User Configuration window displays.

Step 2

Click the Presence Viewer for User link in the Service Settings area.

Note

The Presence Viewer for User link will display only if the Home Cluster and Enable User for Unified CM IM and Presence check boxes are checked.

The Presence Viewer displays.

Confidential
Access Levels

Cisco Unified
Communications Manager Administration Considerations

The Confidential
Access Level (CAL) feature is used for restricting calls and other
supplementary features such as transfer, forward, and conferences including
Meet-Me. CAL is a numeric value assigned to any of the following entities:

Device (for
example, an IP Phone)

Line (for
example, a Directory Number)

Trunk (for
example, a SIP trunk)

CAL has two main
functions:

Controls call
completion based on configuration.

Displays
information on the phone that conveys additional information about the call.

Format of CAL Matrix

The Confidential
Access Level (CAL) matrix is an X/Y matrix that is used to compare one CAL to
another for implementing a call policy. The CAL from the originating number is
selected along the X-axis of the matrix and compared against the destination
number along the Y-axis of the matrix. The intersection of these two values is
known as the resolved CAL. The resolved CAL determines whether the call should
proceed and also the message that is displayed to the users.

A sample CAL
matrix is as follows:

Column 1

Column 2

Column 3

Column 4

Column 5

Description

CAL

1

2

3

Unrestricted

1

1

1

1

Restricted

2

1

2

2

Confidential

3

1

2

3

END

Description

Unrestricted

Restricted

Confidential

Important:

The matrix must
be symmetrical. For example, in the sample CAL matrix above, the value at the
intersection of CAL 2 and CAL 3 is same as the value at the intersection of CAL
3 and CAL 2. Thus, the resolved CAL in both the cases is 2 (Restricted).
Cisco
Unified Communications Manager does not validate if the imported
matrix is symmetrical. So it is the responsibility of the administrator to
configure a matrix that aligns with the desired calling policy.

You can configure
different CALs as per the requirement. The following CALs have been configured
in this sample matrix:

1—Unrestricted

2—Restricted

3—Confidential

The first row of the CAL matrix must contain all the valid CALs that
you want to import into Cisco Unified Communications Manager. Description and
CAL values are optional. The CALs in the remaining columns can be any numeric
values that you want to import. The subsequent rows define the textual
description, as seen in column 1, and its relationship with other CALs in
column 3 and the subsequent columns. For every CAL entered in the first row,
there should be a resulting row that contains a textual description for that
value. In other words, column 1 must contain textual descriptions for all the
CALs that are entered in the first row. The last line (END, Description)
indicates the end of the CAL matrix. The CALs beyond this row are not imported.

If a call is
originated from a number whose CAL is 1 (Unrestricted) to a destination number
whose CAL is 2 (Restricted), the resolved CAL is 1 (the intersection of CAL 1
and CAL 2). Hence, the text corresponding to CAL 1—Unrestricted is displayed on
both the phones. Similarly, if the call is between a Restricted party (with CAL
2) and a Confidential party (with CAL 3), then Restricted (corresponding to the
resolved CAL 2) will be displayed on both the phones. Thus, the CAL matrix
resolves to the highest common value possible between all parties of the call.

The following new
fields have been added in both Phone Configuration and Trunk Configuration
windows:

Configurable RTP
and SRTP Port Ranges

The Configurable
RTP and SRTP port ranges feature allows you to configure RTP and Secure RTP
port ranges used by the software based conference bridges, Media Termination
Points (MTP), Music on Hold (MoH), Annunciator media resources, and SIP
endpoints. Two new service parameters—Start Media Port and Number of Ports have
been added to configure RTP port ranges of IPVMS devices. The configurable port
range is 9051-61000. The Start Media Port and Stop Media Port fields in the SIP
profile are used to configure RTP port ranges for SIP end points. The
configurable port range is 2048-65535.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Configurable Set of Nonpreemptable Numbers

Cisco Unified Communications Manager Administration Considerations

Cisco Unified Communications Manager (Unified Communications Manager), Release 10.0(1) allows you to configure a list of destinations that cannot
be pre-empted so that the calls on these destinations are not disconnected even if a higher precedence call is attempted. You can use this feature for calls to emergency
services so that the emergency calls are not disconnected.

An MLPP Preemption Disabled check box has been added to the Calling Party Transformation Pattern settings. You must check this check box to make the numbers in a transformation pattern nonpreemptable.

Bulk Administration Considerations

No changes.

CDR/CAR Considerations

No changes.

IP Phones Considerations

No changes.

RTMT Considerations

No changes.

Security Considerations

No changes.

Serviceability Considerations

You can allow phone presence from a Cisco Unified Communications Manager that is outside of the IM and Presence Service cluster. Default requests from a Cisco Unified Communications Manager that is outside of the cluster will not be accepted by IM and Presence Service. You can also configure a SIP Trunk on Cisco Unified Communications Manager.

You must configure the TLS context before you configure the TLS peer subject.

Computer Telephony
Integration Support for Cluster-Wide Call Park

For cluster-wide
call park, if a cluster node becomes out of service while a call is parked, the
monitored line generates a Call Disconnected event from that node. If all the
nodes in the cluster become out of service, the monitored line generates a
LineOutOfService event. The Parked line remains in service as long as there is
one active node in the cluster.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Computer Telephony
Integration Video Support

Cisco Unified
Communications Manager Administration Considerations

Computer Telephony
Integration (CTI) provides an interface into
Cisco Unified Communications Manager (Unified Communications Manager) that allows
applications to control and monitor calls, devices, and features. The CTI
interfaces are JTAPI and TAPI. Cisco JTAPI is a java library that resides on
the same platform as the JTAPI application. TAPI is a Microsoft Windows
interface that utilizes the Cisco TSP to access Cisco Unified Communications
Manager. Each CTIManager encapsulates details of the cluster from applications
allowing applications access to CTI functionality for the entire cluster
through a single CTI connection.

CTI Video
Support for TAPI

The Video
Capabilities and Multi-Media Information feature allows the TAPI Application to
detect the multimedia capabilities of Line Devices. This information helps the
applications monitoring devices to answer or route video calls to video capable
devices. It also detects a device with a built-in camera from an audio-only
device.

This application
can determine the video capability of the device, the number of screens on a
device, and if the device supports interoperability with telepresence devices.

If the application
is monitoring only calling devices, then called device multimedia capabilities
are communicated after the call is answered. If the application is monitoring
only called devices, then calling device multimedia capabilities are
communicated before the call is answered (for example, when a call is offered).

TAPI provides
video capability information for same cluster calls involved in the following
features:

Basic Call and
Consult Call

Redirect

Call Forward

Hold and
Resume

Hunt List

Transfer

Extension
Mobility

Super Provider

TAPI provides
video capability information for across-cluster calls involved in the following
features:

Basic Call and
Consult Call

Redirect

Call Forward

Hold and
Resume

Hunt List

Extension
Mobility

Super Provider

The multimedia
capability of the device is exposed as a structure DeviceMultiMediaCapability
in the DevSpecific part. This structure contains three fields:

DeviceVideoCapability provides the type value defined in the
enumeration [CiscoDeviceVideoCapabilityInfo].

TelepresenceInfo indicates if Telepresence is enabled on the
device, defined in the enumeration [CiscoDeviceTelepresenceInfo].

ScreenCount
indicates the number of screens present on the device.

Note

The initial
video capability is not supported for CTI Route Points and CTI Ports; however,
they can receive video information.

The following
table describes the video capabilities provided by Cisco TAPI for currently
supported devices.

Table 11 Video
Capability for IP Phones for TAPI

Phone
Model

Protocol

Video
Capability – Registration update to Apps

Dynamic
Video Capability change Notification

Supports
TIP & Screen Count

Does CTI
send MultiMedia streams Notification Event to applications?

8945

SCCP

Yes

Yes

No

No

8945

SIP

Yes

Yes

No

Yes

6921/6941

SCCP

Yes

Yes

No

No

9971/9951

SIP

Yes

Yes

No

Yes

EX90

SIP

Yes

N/A

No

Yes

CTIPort

SCCP

Yes
[Video disabled]

N/A

N/A

No

CTIRoutePoint

SCCP

Yes
[Video disabled]

N/A

N/A

No

CTS 500

CTS
500-32

SIP

Yes

N/A

Yes

Yes

Jabber (CSF/softphone mode)

SIP

Yes

Yes

Screen
Count-No

TIP-No

Yes

CTI Video
Support for JTAPI

In
Unified Communications
Manager, Release 10.0(1), JTAPI is exposing video capabilities for
supported terminals and calls. Video capabilities for near and far-end
terminals include whether they are video-enabled, inter-operability with
TelePresence, and the number of screens. Video attributes for calls will also
be available to JTAPI applications which would include IP/port address, codec,
and other information. Using the provided video terminal and call information,
JTAPI applications will be able to better handle calls like routing incoming
video-capable calls to agents with video-enabled terminals.

Exposing MultiMedia
Capability on CiscoTerminal: Cisco JTAPI provides a new API,
getCiscoMultiMediaCapabilityInfo() on Cisco Terminal to expose the multimedia
capabilities of the terminal. These capabilities are exposed on a new interface
CiscoMultiMediaCapabilityInfo, which will have the following APIs to expose
these capabilities:

getVideoCapability()

getTelepresenceInfo()

getScreenCount()

Exposing changes in
MultiMedia Capability via a new provider event: Any change in video
capability of the terminal will be notified to the application by a new JTAPI
event (CiscoProvTerminalMultiMediaCapabilityChangedEv). Video capability can be
changed only from the Admin Device Configuration pages. Plugging in or out a
Cisco Camera does not affect the video capability status, hence the new event
is not triggered in this case. This event is a JTAPI provider event, and will
be delivered only if the application has added provider observers. The terminal
has to be in the registered state as a pre-condition for receiving this event.

Note

A change in
Multimedia Capability through CiscoProvTerminalMultiMediaCapabilityChangedEv
will not be delivered to applications when the video capability of an SCCP
Phone changes. In this case, the terminal will unregister and register back;
therefore the application needs to update the video capability after the
terminal is registered.

Exposing MultiMedia
Capability on a CiscoCall: An application can detect if the far-end Party
for an incoming call is video capable prior to media setup. Consider a scenario
where A calls B, the multimedia capabilities of the calling and called party
will be exposed on the CiscoCall on terminal B after the call is offered to
terminal B. The Cisco JTAPI provides the
getCallingTerminalMultiMediaCapabilityInfo () and
getCalledTerminalMultiMediaCapabilityInfo() APIs on the CiscoCall to expose the
multimedia capabilities of the calling and called party in a call.

The same APIs
can be used to determine the multimedia capabilities for an outgoing call, but
note that the video capability will be known only after the call is answered.
Consider a scenario where A calls B, B answers the call, the multimedia
capabilities of the calling and called party will be exposed on the CiscoCall
on terminal A after the call is answered by terminal B. The APIs
getCallingTerminalMultiMediaCapabilityInfo() and
getCalledTerminalMultiMediaCapabilityInfo() return
CiscoMultiMediaCapabilityInfo.

Exposing MultiMedia
Streams Information on Cisco Terminal: The new JTAPI terminal event
CiscoMultiMediaStreamsInfoEv will be delivered to a terminal observer to
indicate multimedia streams information of a call. The multimedia streams
information is exposed on the interface CiscoMultiMediaProperties, via the API
getProperties() on CiscoMultiMediaStreamsInfoEv. The Cisco JTAPI provides the
multimedia streams information of the terminal after a call is connected. A
MultiMedia Stream may include a video stream, a presentation stream, or both.

A video capable
device is a device that can perform any of the following functions:

JTAPI will
provide video capability information for same cluster calls involved in the
following features:

Originating
Call and Consult Call

Redirect

Call Forward

Hold and
Resume

Hunt List

Transfer

Extension
Mobility

Super
Provider

JTAPI will
provide video capability information for across-cluster calls involved in the
following features:

Basic Call
and Consult Call

Redirect

Call Forward

Hold and
Resume

Hunt List

Extension
Mobility

Super
Provider

The following
table describes the video capabilities provided by Cisco JTAPI for currently
supported devices.

Table 12 Video
Capability for IP Phones for JTAPI

Phone
Model

Protocol

Support
Initial Device Multimedia Capability on Cisco Terminal

Supports
Multimedia Capabilities on Cisco Call

Supports
Multimedia Streams Information

Dynamic
Video Capability Change

8945

SCCP

Yes

Yes

No

Yes

8945

SIP

Yes

Yes

Yes

Yes

9951/9971

SIP

Yes

Yes

Yes

Yes

EX60/90

SIP

Yes

Yes

Yes

N/A

CTIPort

SCCP

N/A

Yes

No

N/A

CTIRoutePoint

SCCP

N/A

Yes

No

N/A

CTS
500-32

SIP

Yes

Yes

Yes

N/A

Jabber
(CSF/softphone mode)

SIP

Yes

Yes

Yes

No

Limitations
for TAPI and JTAPI

The following
are the limitations of the Video Capabilities and Multi-Media Information
feature for TAPI.

Remote in
Use - CiscoTSP does not provide correct calling and called party multimedia
capabilities on a call that is in inactive state or is in Remote InUse state.

MultiMedia
Capability Information:

Calling
and called party multimedia capabilities will be UKNOWN on the calling side
until the called party answers the call.

If an
outbound call is initiated over SIP Trunk configured with Early Offer then the
called party will just respond back with the capabilities it was offered during
the initial offer and not its complete capabilities.

Only
video capability information will be known for calls over H323 trunk, Screen
count and telepresence interoperability information will be unknown.

MultiMediaStreams Information - CiscoTSP does not provide
multimedia streams information if the device is a SCCP phone; therefore,
CiscoTSP will not deliver SLDSMT_MULTIMEDIA_STREAMSDATA and
TSPI_LineGetCallInfo() API will not provide multimedia streams information in
VideoStreamInfo structure.

Change in
called party - In scenarios like Shared Lines or redirect, where the called
party changes, the application will be notified of the new called party
capability only if they configure the called party with unique display names.

The following
are the limitations of the Video Capabilities and Multi-Media Information
feature for JTAPI.

Outgoing
call - Applications observing only calling party will have calling and called
party multimedia capabilities as UKNOWN until the called party answers the
call.

Shared Line
- Incoming call - calling and called party multimedia capabilities only if at
least one of the terminal connections on the cisco call is not in passive
state.

Shared Line
- Incoming Call - Called party multimedia capabilities will not have correct
multimedia capabilities when more than one terminal connection is in ringing
state.

MultiMedia
Streams Information - Cisco JTAPI will not deliver CiscoMultiMediaStreamsInfoEv
on a CiscoTerminal which is a SCCP phone.

Incoming
Call - If an outbound call is initiated over SIP Trunk configured with Early
Offer then the called party will just respond back with the capabilities it was
offered during the initial offer and not its complete capabilities.

Change in
called party - In scenarios like Shared Lines or redirect, where the called
party changes, the application will be notified of the new called party
capability only if they configure the called party with unique display names.

Bulk
Administration Considerations

CDR/CAR
Considerations

No Changes.

IP Phones
Considerations

No Changes.

RTMT
Considerations

No Changes.

Security
Considerations

No Changes.

Serviceability
Considerations

No Changes.

Dial-Via-Office
Reverse Voicemail Policy

This feature
configures how dual mode device users answer Dial-via-Office Reverse (DVO-R)
calls that terminate on the Mobile Identity (MI). This feature provides users
with a single enterprise voicemail box for their enterprise mobility if the RD
call reaches an external voice mail system. Available options are as follows:

Use System
Default

Timer Control

User Control

Cisco Unified
Communications Manager Administration Considerations

On the
Remote
Destination settings window, the following drop-down list box has
been added: Dial-via-Office Reverse Voicemail Policy.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

This feature
enables Location-based Call Admission Control (CAC) for Cisco Extension
Mobility Cross Cluster (EMCC) calls to be based on the usual location
configuration of the EMCC phone for the Enhanced Location Call Admission
Control feature.

Prior to release
10.0(1), when an EMCC phone was involved in a call after it registered with the
home cluster, Cisco Unified Communication Manager (Unified Communications
Manager), used the local EMCC location configured with the roaming device pool
on the home cluster Unified Communications Manager for both RSVP and static
Location-based CAC.

With release
10.0(1), when an EMCC phone registers with the home cluster Unified
Communications Manager , the location configuration for the physical phone on
the visiting cluster is passed to the home cluster. If the home cluster
supports the Enhanced Location CAC feature and participates in replication with
the visiting cluster Location Bandwidth Manager (LBM) service, the home cluster
uses the visiting cluster location for the location CAC calculation for the
EMCC phone. RSVP CAC keeps using the home cluster roaming device pool location
since RSVP policy can only be configured on intra-cluster location pairs. If
the LBM on the home cluster does not connect to an LBM on the visiting cluster
(Enhanced Location CAC is not enabled, or Enhanced Location CAC replication is
not setup between the home cluster and the visiting cluster), the home cluster
roaming device pool location is used to keep the existing pre-10.0(1) Unified
Communications Manager behavior.

When a Cisco
CallManager service connects with an LBM service, the LBM service sends the
list of all the remote LBM clusters in the same LBM replication network to the
Cisco CallManager service. LBM service also sends the update to Cisco
CallManager service if the LBM replication network has any change. When an end
user from home cluster logs into the EMCC phone on a visiting cluster,
extension mobility service on the visiting cluster sends the location
configuration for the EMCC phone and the visiting cluster's ID to extension
mobility service on home cluster to save into the database on home cluster.
When the EMCC phone registers with the Cisco CallManager service on the home
cluster after login succeeds, Cisco CallManager service checks whether the
visiting cluster ID for the EMCC phone is in the LBM replication network. If
yes, the Cisco CallManager service uses the location configuration from the
visiting cluster for the EMCC phone. Otherwise, the Cisco CallManager service
uses the home cluster roaming device pool location for the EMCC phone.

EMCC with
Different Releases of a Cisco Unified Communication Manager

The EMCC feature
was introduced with Unified Communications Manager Release 8.0 and the Enhanced
Location CAC feature was introduced with Unified Communications Manager Release
9.0. The EMCC feature can be configured between the Unified Communications
Manager clusters with the same or different releases.

The following
table lists the EMCC behavior with different releases of Unified Communications
Manager.

Table 13 EMCC Behavior
with Different Releases

Home
Cluster

Visiting Cluster

Enhanced Location CAC
Enabled Between Clusters

EMCC phone location in
home cluster behavior

9.0 or
earlier release

9.0 or earlier release

Not Supported

Uses default home cluster
roaming device pool location

9.0 or
earlier release

10.0 or later release

Not Supported

Uses default home cluster
roaming device pool location

10.0 or
later release

9.0 or earlier release

Not Supported

Uses default home cluster
roaming device pool location

10.0 or
later release

10.0 or later release

Yes

Uses visiting cluster
location configured for device

10.0 or
later release

10.0 or later release

No

Uses default home cluster
roaming device pool location

Restrictions
and Interactions

If the visiting
cluster administrator changes the location assigned to the EMCC device after
the EMCC device on visiting cluster is logged in by a remote user from the home
cluster, the change does not affect the EMCC device location in the home
cluster until the user logs out and the same or another user logs in the EMCC
device.

If the
administrator changes the name of the location used by the EMCC device after
the EMCC device on the visiting cluster is logged in by a remote user from the
home cluster, LBM on the home cluster gets the name change propagation from the
visiting cluster while Cisco CallManager service on the home cluster still has
the old location name for the EMCC device. When the EMCC device makes a call,
the bandwidth reservation will fail with no path error until the user logs out
and the same or another user logs in the EMCC device.

If the LBM
services communication is lost between the home cluster and the visiting
cluster and LBM service on the home cluster does not recognize the remote
location from the visiting cluster after EMCC device is logged in, the call
reservation will follow the Unified Communications Manager Release 9.0 Enhanced
Location CAC error condition behavior.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Flexible DSCP Marking and Video Promotion

Devices and applications use Differentiated Services Code Point (DSCP) markings to indicate the Quality of Service (QoS) treatment of IP communications. For example, desktop video endpoints may use multimedia conferencing AF41 marking for video media streams, while high-definition video room systems may use real-time interactive CS4 marking. When an application sends and receives IP communications to and from the same type of application, the DSCP markings are symmetric, and the QoS treatments of the IP communications that each application sends and receives are the same. However, when an application sends and receives media to and from a different type of application, the DSCP markings may be asymmetric, and the QoS treatments of the IP communications that each application sends and receives may be inconsistent. For example, the QoS treatment of the video media stream that a video room system receives from a desktop video endpoint may be inadequate to support the expected quality of the video room system.

Devices and applications are subjected to Call Admission Control (CAC) to ensure that adequate bandwidth is available for the duration of established sessions. The bandwidth that is utilized by established sessions is updated as the sessions begin and end. Attempts to establish new sessions that would exceed the available bandwidth are blocked. The amount of bandwidth available may be tracked independently for devices and applications of different types. For example, independent tracking of bandwidth may be available for desktop video endpoints and high-definition video room systems to send and receive video media streams.

When devices and applications of the same type send and receive communications to and from each other, the same type of bandwidth deductions are made in each direction. However, when devices and applications of different types send and receive communications to and from each other, different types of bandwidth deductions must be made in each direction. Moreover, the bandwidth deductions are usually symmetric in amount, by design, to reflect the usual behavior of an IP network. As a result, when devices and applications of different types send and receive communications to and from each other, the total bandwidth deductions may be up to double the amount of network bandwidth that is actually utilized. This inconsistency in bandwidth accounting may cause attempts to establish new sessions to be blocked unnecessarily. For example, when a desktop video endpoint and a Cisco TelePresence immersive video endpoint are in a call, in Release 9.x Unified Communications Manager CAC design deducts the same amount of bandwidth in both the video bandwidth pool and in the immersive bandwidth pool, because these two video endpoints are marking DSCP differently and the media packets can potentially traverse in different queues. This behavior unnecessarily deducts double the bandwidth that is required and potentially blocks new video calls.

In Unified Communications Manager Release 10.0(1) and later releases, the system administrator can configure a Video Promotion policy that reconciles the inconsistency in bandwidth accounting in favor of the application that receives more favorable CAC and QoS treatment. For example, if a session between a desktop video endpoint and a high-definition video room system is reconciled in favor of the video room system, then the reconciliation is deemed a promotion for the desktop video endpoint.

When reconciliation is in effect between devices and applications of different types, bandwidth is deducted only for the type of application that is favored by reconciliation. If sufficient bandwidth is available for a session of this type to be admitted, the device or application of the type that is not favored by reconciliation is instructed to change the DSCP markings that it uses to those that are used by the device or application of the type that is favored by reconciliation.

For example, if a desktop video endpoint is promoted in a session with a high-definition video room system, bandwidth accounting takes place as if the desktop video endpoint were an application of the same type as the video room system. The desktop video endpoint is instructed to change its DSCP markings to those that are used by the video room system. The QoS treatment is consistent in both directions, bandwidth is deducted for a session between devices and applications of the same type as the video room system, and bandwidth is not deducted for a session between devices and applications of the same type as the desktop video endpoint.

To activate the Flexible DSCP Marking and Video Promotion feature, in the Service Parameter Configuration window set the Use Video BandwidthPool for Immersive Video Calls service parameter to False and set the Video Call QoS Marking Policy service parameter to Promote to Immersive. When the Flexible DSCP Marking and Video Promotion feature is activated, Unified Communications Manager dynamically signals desktop video devices a Traffic Class Label that is indicative of the DSCP marking for each negotiated media stream.

Traffic Class Label

The Flexible DSCP and Video Promotion feature uses the Traffic Class Label (TCL) to instruct the SIP endpoint dynamically to mark its DSCP on a per call basis, based on the Video Promotion policy that is defined by the system administrator. Because TCL is a SIP Session Description Protocol (SDP) attribute that is defined per media line, the TCL and its associated DSCP markings can be different for the audio media line and the video media line of a video call. The system administrator can choose different DSCP markings for the audio stream and the video stream of the video call.

Interactions and Restrictions

The following interactions and restrictions apply to the Flexible DSCP Marking and Video Promotion feature:

The Flexible DSCP Marking and Video Promotion feature is dependent on desktop SIP video endpoint support. At the time of the initial release of Unified Communications Manager Release 10.0(1), only Cisco DX650 series SIP phones provide the required endpoint support.

If pass-through MTPs are inserted in a call, Unified Communications Manager signals the MTP to mark the packets with the DSCP marking that is expected from the endpoint device that originally emitted the packet for the video stream. If the two endpoints on a call use different DSCP markings (for example, a Cisco TelePresence immersive video endpoint and a desktop video endpoint without Video Promotion), the MTPs preserve the DSCP marking in each stream direction.

Cisco recommends that you do not use the Flexible DSCP Marking and Video Promotion feature with Multilevel Precedence and Preemption (MLPP) service calls. When you need MLPP service functionality, Cisco recommends that you set the Video Call QoS Marking Policy and Use Video BandwidthPool for Immersive Video Calls service parameters to their default values. With default values for the Video Call QoS Marking Policy and Use Video BandwidthPool for Immersive Video Calls service parameters, Unified Communications Manager and endpoints use MLPP DSCP markings for the media packets.

Service Parameters for Flexible DSCP Marking and Video Promotion

Unified Communications Manager Release 10.0(1), and later releases, provides the following clusterwide service parameters to configure the Flexible DSCP Marking and Video Promotion feature:

Video Call QoS Marking Policy. This parameter allows the administrator to configure a Promote to Immersive policy that reconciles bandwidth allocation inconsistencies between a desktop video endpoint and a Cisco TelePresence immersive video endpoint in favor of the immersive endpoint. When promotion is performed, the audio and video bandwidth are reserved from the immersive bandwidth pool allocation. The policy of Promote to Immersive takes effect only for calls between an immersive video device and a desktop video device that supports flexible DSCP marking.
To configure a Promote to Immersive policy, in the Service Parameter Configuration window set the Use Video BandwidthPool for Immersive Video Calls parameter to False and set the Video Call QoS Marking Policy parameter to Promote to Immersive.

DSCP for Video Calls. This parameter specifies the DSCP value for the video stream of video calls.

DSCP for Audio Portion of Video Calls. This parameter specifies the DSCP value for the audio stream of video calls.

DSCP for TelePresence Calls. This parameter specifies the DSCP value for the video stream of Cisco TelePresence video calls.

DSCP for Audio Portion of TelePresence Calls. This parameter specifies the DSCP value for the audio stream of Cisco TelePresence video calls.

Default Intraregion Max Immersive Video Call Bit Rate (Includes Audio). This parameter specifies the default maximum total bit rate for each immersive video call within a particular region, when the Use System Default option is selected as the Max Immersive Video Call Bit Rate in the Region Configuration window for the relationship of the region with itself. For more information about choosing the options in the Region Configuration window, see the Cisco Unified Communications Manager Administration Guide.

Default Interregion Max Immersive Video Call Bit Rate (Includes Audio). This parameter specifies the default maximum total bit rate for each immersive video call between a particular region and another region, when the Use System Default option is selected as the Max Immersive Video Call Bit Rate in the Region Configuration window for the relationship of the region with the other region. For more information about choosing the options in the Region Configuration window, see the Cisco Unified Communications Manager Administration Guide.

Additional
Information

Unified Communications Manager Release 10.0(1), and
later releases, provides eight clusterwide service parameters to configure the
Flexible DSCP Marking and Video Promotion feature. The following five new
parameters were introduced in Release 10.0(1):

The following
three parameters, which are also required to configure the Flexible DSCP
Marking and Video Promotion feature, were introduced prior to Release 10.0(1):

DSCP for Video
Calls

DSCP for
TelePresence Calls

Use Video
BandwidthPool for Immersive Video Calls

Cisco Unified
Communications Manager Administration Considerations

The following
table describes two new fields that have been added to the
Region
Configuration window for the Flexible DSCP Marking and Video
Promotion feature.

Table 14 Region
Configuration Settings

Field

Description

Region
Relationships

Maximum
Session Bit Rate for Immersive Video Calls

The
entries in this column specify the maximum immersive video bit rate (including
audio) between the region that you are configuring and the region that displays
in the corresponding row.

Modify
Relationship to other Regions

Maximum
Session Bit Rate for Immersive Video Calls

For each
region that is specified in the Regions window pane, click one radio button in
this column as specified:

Keep
Current Setting—Click this button to use the current setting for the immersive
video call bandwidth.

Use
System Default—Click this button to use the default value. The default value
normally specifies 2000000000 kbps, unless the default value has been set to a
different value in the Service Parameters Configuration window.

None—Click this radio button if no immersive video call bit rate
is allotted between this region and the specified region. If you choose this
option, the system does not allow immersive video calls.

kbps—Click this button to set the maximum immersive video call
bitrate between the region that you are configuring and the specified region.
Enter the bit rate that is available for each immersive video call between
these two regions; remember that the audio bit rate is included. Valid values
range from 1 to 2147483647.

From the Server drop-down list, choose the server where you want to configure the parameters.

Step 3

From the Service drop-down list, choose the Cisco CallManager (Active) service.

If the service does not display as active, ensure that the service is activated in Cisco Unified Serviceability.

Step 4

To configure the parameters, scroll to the appropriate area of the Service Parameter Configuration window and update the parameter values.

Note

To configure a Video Promotion policy that promotes desktop video endpoints to immersive video endpoints, set the Use Video BandwidthPool for Immersive Video Calls parameter to False and set the Video Call QoS Marking Policy parameter to Promote to Immersive.

Cisco Unified
Communications Manager Administration Considerations

Bulk
Administration Considerations

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Global Dial Plan
Replication

Global Dial Plan
Replication allows you to create a global dial plan that spans across an ILS
network and which includes intercluster dialing of directory URIs and alternate
numbers. When Global Dial Plan Replication is enabled, the Intercluster Lookup
Service (ILS) advertises global dial plan data, including locally configured
and any data that was learned from other clusters, to the ILS network. Global
dial plan data includes the following:

Directory
URIs

Alternate
Numbers

Advertised
Patterns

PSTN
Failover

Route String

Learned
Global Dial Plan Data

Imported
Global Dial Plan Data

Directory
URIs

ILS replicates
the full catalog of locally configured directory URIs where the Advertise
Globally via ILS option has been selected. The
"URI
dialing" chapter in the
Cisco
Unified Communications Manager Features and Services Guide contains
details on how to set up URI dialing.

Alternate
Numbers

Alternate
numbers can be configured as aliases of directory numbers. Alternate numbers
allow you to configure globally routable numbers that can be dialed from
anywhere within an ILS network.
Cisco Unified Communications Manager (Unified Communications Manager) allows you to create
two types of alternate numbers:

Enterprise
alternate numbers

+E.164
alternate numbers

In Cisco Unified
Communications Manager Administration, you can create an enterprise alternate
number and an +E.164 alternate number and associate both alternate numbers to a
directory number. When you associate an alternate number to a directory number,
the alternate number can act as an alias of that directory number so that when
you dial the alternate number, the phone that is registered to the associated
directory number rings.

Each alternate
number that you set up must associate to a single directory number. However,
that directory number can associate to both an enterprise alternate number and
an +E.164 alternate number at the same time. You can also choose one of the
alternate numbers as the PSTN failover number for all alternate numbers and
directory URIs that are associated to that directory number.

Advertised
Patterns

Advertised
patterns allow you to create summarized routing instructions for a range of
enterprise alternate numbers or +E.164 alternate numbers and replicate that
pattern throughout an ILS network such that all clusters within the ILS network
know the pattern. Advertised patterns save you from having to configure routing
information for each alternate number on an individual basis. Advertised
patterns are never used by the local cluster on which they are configured; they
are only used by remote clusters that learn the pattern through ILS.

For example, if
Cluster A has a range of enterprise alternate numbers between 80001-89999 and
you want to replicate those alternate numbers throughout the ILS network, you
can create a pattern of 8XXXX and advertise that pattern to the ILS network.
When a remote cluster receives an outgoing call for which the dial string
matches the learned pattern (for example, 82211), the remote cluster uses the
route string that is associated with the pattern to route the call.

PSTN
Failover

When Global Dial
Plan Replication is enabled, ILS can be configured to replicate a PSTN failover
rule for learned directory URIs, learned numbers, and learned patterns. If the
dial string for an outgoing call matches a learned pattern, learned alternate
number, or learned directory URI, and
Unified Communications
Manager is unable to route the call over a SIP trunk,
Unified Communications
Manager uses the calling party's AAR CSS to reroute the call to the
associated PSTN failover number.

Unified Communications Manager uses the PSTN failover
for routing only for calls placed to learned patterns, learned alternate
numbers, or learned directory URIs.
Unified Communications
Manager does not route calls to the PSTN failover number for calls
that are placed to patterns, alternate numbers, or directory URIs that were
configured in the local cluster.

Route
Strings

To configure
Global Dial Plan Replication, you must assign a distinct route string for each
cluster in the ILS network. Route strings can be up to 250 alphanumeric
characters, including dots (.) and dashes(-). Although route strings are used
with domain-based routing, route strings do not have to match a specific
domain; you can assign whatever route strings you want.

When you assign
a route string to a cluster, ILS associates that route string to all the global
dial plan data that is local to that cluster (including locally configured
directory URIs, alternate numbers, advertised patterns, and PSTN failover
information). If Global Dial Plan Replication is enabled, ILS advertises the
local route string and the rest of the global dial plan data to the ILS
network.

To configure
remote
Unified Communications
Manager clusters to route to the route string, for each cluster in
the ILS network, you must configure SIP route patterns that match the route
strings in the ILS network and route calls that are destined for those route
strings to SIP trunks that lead to the next-hop clusters in your ILS network.

When a user in a
remote cluster dials a directory URI or alternate number that was learned via
ILS,
Unified Communications
Manager pulls the associated route string, matches that route string
to a SIP route pattern, and routes the call to the trunk that is specified by
the SIP route pattern.

Imported
Global Dial Plan Data

ILS also
advertises global dial plan data that has been imported from a CSV file into
any hub cluster in the ILS network. Imported global dial plan data includes
directory URI catalogs, +E.164 patterns, and PSTN failover numbers for a call
control system that does not run ILS, such as a Cisco TelePresence Video
Communication Server, or a third-party call control system.

Cisco Unified
Communications Manager Administration Considerations

Call
Routing > Directory Number: The Directory Number
Configuration window has the following updates:

An
Enterprise Alternate Number section has been added that allows you to create an
enterprise alternate number as an alias of the directory number and advertise
that alternate number to the ILS network.

An +E.164
Alternate Number section has been added with the same fields and capability as
with Enterprise Alternate Numbers.

The
Directory URIs section has been updated with a check box that you must check
for ILS to replicate the directory URI to remote clusters.

A PSTN
failover option has been added where you can choose one of the alternate
numbers as a PSTN failover for all the alternate numbers and directory URIs
that are associated to this directory number.

Call
Routing > Global Dial Plan Replication > Advertised
Patterns: This is a new menu item that allows you to
create an alternate number pattern that ILS advertises to the ILS network.
Advertised patterns allow you to create a single number pattern that summarizes
a range of alternate numbers.

Call
Routing > Global Dial Plan Replication > Block Learned Numbers and
Patterns: This is a new menu item that allows you to
create a blocking rule for alternate numbers or patterns that have been learned
with ILS.

Call
Routing > Global Dial Plan Replication > Partitions for Learned
Numbers and Patterns: This is a new menu item that
allows you to assign alternate numbers and alternate number patterns that have
been learned with ILS to a partition on the local cluster.

Call
Routing > Global Dial Plan Replication > Learned
Numbers: This is a new menu item that allows you to
view all of the alternate numbers that the local cluster has learned with ILS.

Call
Routing > Global Dial Plan Replication > Learned
Patterns: This is a new menu item that allows you to
view all of the number patterns that the local cluster has learned with ILS.

Call
Routing > Global Dial Plan Replication > Learned Directory
URIs: This is a new menu item that allows you to view
all of the directory URIs that the local cluster has learned with ILS.

Bulk
Administration > Directory URIs and Patterns > Export Local Directory
URIs and Patterns: This is a new menu item that
allows you to export directory URIs, +E.164 advertised patterns, and PSTN
failover rules to a CSV file.

The following
line fields have been added to the
bat.xlt
import spreadsheet:

Enterprise
Is Urgent

Enterprise
Add to Local Route Partition

Enterprise
Advertise Via Globally

Enterprise
Number Mask

Enterprise
Route Partition

E.164 Is
Urgent

E.164 Add to
Local Route Partition

E.164
Advertise Via Globally

E.164 Number
Mask

E.164 Route
Partition

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

Set Up Global Dial
Plan Replication

This procedure
describes how to set up Global Dial Plan Replication in the ILS network. See
the Related Topics for more detailed information on how to perform some of the
high-level steps in this procedure.

Before You Begin

Global Dial Plan
Replication runs on an ILS network. Follow the procedure to set up an ILS
network in the
"Intercluster
Lookup Service" chapter before you configure Global Dial Plan Replication.

Procedure

Step 1

Enable ILS
support for Global Dial Plan Replication in the local cluster:

(Optional) If
you want to set up a PSTN failover number for specific directory URIs or
alternate numbers, assign an alternate number as the PSTN failover number for
all the directory URIs and alternate numbers that are associated to a specific
directory number.

Step 5

(Optional) If
you want to summarize your alternate numbers with a pattern, set up an
advertised pattern, and assign a PSTN failover rule for the pattern.

Step 6

In the
Partitions for Learned Numbers and Patterns
configuration window, assign route partitions to the alternate numbers and
patterns that the local cluster learns through ILS.

Step 7

Set up SIP
route patterns to route calls to the remote clusters in your ILS network by
doing the following:

Create SIP
route patterns that match the route strings for the remote clusters in the ILS
network.

Point
those SIP route patterns to SIP trunks or route lists that route calls to the
next-hop clusters in the ILS network.

Step 8

If your
network includes a Cisco Unified Border Element, do the following for the SIP
profiles in your network:

Set an upper
limit for the number of learned objects that ILS can write to the local
database by setting a value for the ILS Max Number of Learned Objects service
parameter. The default value is 100,000.

Step 10

Repeat the
previous steps for each cluster in your ILS network.

Step 11

(Optional) If
you want your ILS network to interoperate with a Cisco TelePresence Video
Communication Server or third-party call control system, import directory URI
catalogs from a CSV file for the other system into any hub cluster in the ILS
network.

Find and
select the directory number to which you want to associate the alternate
number.

Step 3

Click either
Add
Enterprise Alternate Number or
Add
+E.164 Alternate Number depending on which type of alternate number
you want to assign.

Step 4

In the
Number Mask field, enter the number mask that
you want to apply to the directory number. The
Alternate Number field displays how the
alternate number appears after
Cisco
Unified Communications Manager applies the number mask.

Step 5

(Optional) If
you want to enable local routing for the alternate number, do the following:

Check the
Add to Local Route Partition check box.

From the
Route Partition drop-down list box, choose
a route partition that is assigned to a local calling search space.

Step 6

(Optional) If
you want to use a number pattern to set up intercluster routing for this
alternate number, click
Save and end the procedure. See the Related Topics
section for a procedure on how to advertise an alternate number pattern to the
ILS network.

Step 7

(Optional) If
you want to set up intercluster routing for this alternate number, check the
Advertise Globally via ILS check box for this
alternate number.

Step 8

(Optional) If
you want to assign a PSTN failover number to this alternate number, from the
PSTN failover drop-down list box, assign a
number as the PSTN failover.

Step 9

Click
Save.

What to Do Next

If you want to
enable intercluster routing for the alternate number you must also set up
Global Dial Plan Replication within your ILS network. ILS will not advertise
the alternate number unless Global Dial Plan Replication is enabled.

Set Up PSTN
Failover for Directory URIs and Alternate Numbers

This procedure
describes how to assign a PSTN failover number for directory URIs or alternate
numbers and advertise that PSTN failover number to the ILS network. Remote
clusters can use the PSTN failover number for calls to learned directory URIs
or learned alternate numbers.

Find and
select the directory number that is associated to the directory URI or
alternate number for which you want to assign a PSTN failover number.

Step 3

If the
alternate number that you want to use as the PSTN failover does not exist,
create either an enterprise alternate number or a +E.164 alternate number for
the directory number.

Step 4

In the PSTN
Failover drop-down list box, choose the alternate number that you want to use
as the PSTN failover.

Step 5

Click
Save.

Cisco Unified
Communications Manager associates that PSTN failover number to that directory
number. Global Dial Plan Replication advertises that number to the ILS network
as the PSTN failover number for all the directory URIs and alternate numbers
that are associated to that directory number.

What to Do Next

In order for a
remote cluster to route calls to the PSTN failover number, you must set up the
AAR CSS and configure route patterns in the remote cluster that route the PSTN
failover number to a PSTN gateway.

Set Up an Advertised Pattern for Alternate Numbers

Follow this
procedure to create a pattern that summarizes a range of alternate numbers and
advertise the pattern to the ILS network.

In the
Pattern field, enter the pattern that you want
to advertise to the ILS network.

Step 4

Use the
Pattern Type radio buttons to choose whether
you want to apply the pattern to a range of enterprise alternate numbers or
+E.164 alternate numbers.

Step 5

Complete the
remaining fields in the
Advertised Patterns Configuration window to
configure a PSTN failover rule for the pattern.

Step 6

Click
Save.

If Global Dial
Plan Replication is enabled, ILS advertises the pattern to remote clusters in
the ILS network.

What to Do Next

For remote
clusters to be able to route calls to the PSTN failover number, in the remote
cluster you must set up AAR and create route patterns that route the PSTN
failover digits to a PSTN gateway.

Block a Learned
Pattern

If you want to
prevent a local
Cisco
Unified Communications Manager cluster from routing calls to a
learned alternate number or learned alternate number pattern, you can configure
a local blocking rule on that cluster. Before routing a call to a learned
number or learned pattern, ILS checks to see if a local blocking rule matches
the dial string. If the blocking rule matches,
Cisco
Unified Communications Manager does not route the call.

Some additional
characteristics of blocking rules:

Blocking rules
are applied only on the local cluster on which you configure them—ILS does not
advertise blocking rules.

In the Blocked
Pattern section, complete the fields that you want to use as conditions for the
blocking rule. If you do not want to use a specific field as a blocking
condition, you can leave that field blank. For example:

If you want to block all
calls to ABC_cluster1 regardless of the other call parameters, enter
ABC_cluster1 in the
Cluster ID field, click the
Any radio button, and leave the remaining fields
empty.

If you want to block all
+E.164 calls to Cluster_3 that use a prefix of 683, enter “Cluster_3” in the
Cluster ID field, enter “683” in the Prefix field, click the
+E.164 Pattern radio button, and leave the remaining
fields empty.

If you want to block a
specific enterprise pattern, enter the pattern in the Pattern field and click
the
Enterprise Pattern radio button.

Step 4

In the
Pattern type field, choose whether you want to apply the blocking rule to
Enterprise patterns, +E.164 patterns, or both.

Step 5

Click
Save.

IM and Presence
Service Group Chat and Persistent Chat Configuration

This section
describes how to configure the enhanced ad hoc and persistent chat settings.
These settings are configured with default values that you can modify. You can
revert all settings to their default values by clicking the
Set to Default
button.

Note

To allow chat room
owners to change a setting, check the
Room owners can
change check box on the server. The room owner can then configure such
settings as they wish and those settings are applicable to the room they are
creating. The availability of configuring these settings from the client also
depends on the client implementation and whether the client is providing an
interface in which to configure these settings.

Configure group
chat alias settings

Group chat alias
settings allow users in any domain to search for specific chat rooms on
specific nodes, and join in those chat rooms.

Procedure

Step 1

Check
System
automatically manages primary group chat server aliases if you want
to enable the system to automatically assign chat room aliases to nodes, using
the alias naming convention
"conference-x-clusterid.domain."The check box is checked by
default.

Step 2

Click
Save.

Note

If you are
adding, deleting, or modifying aliases you must restart the Cisco XCP Text
Conference Manager on all nodes in the cluster by selecting
Cisco Unified IM and
Presence Serviceability > Tools > Control Center - Feature
Services.

Enable Persistent
Chat

You need
to configure persistent chat settings only if you use persistent chat rooms as
opposed to temporary (ad hoc) chat rooms. This configuration is specific to
persistent chat and has no impact on IM archiving for regulatory compliance.

Before You Begin

To use persistent chat
rooms, you must configure a unique external database instance for each node.

If you use an external
database for persistent chat logging, consider the size of your database.
Archiving all the messages in a chat room is optional, and will increase
traffic on the node and consume space on the external database disk. In large
deployments, disk space could be quickly consumed. Ensure that your database is
large enough to handle the volume of information.

Archiving all
room joins and leaves is optional, because it increases traffic and consumes
space on the external server.

Before you configure the
number of connections to the external database, consider the number of IMs you
are writing and the overall volume of traffic that results. The number of
connections that you configure will allow the system to scale. While the
default settings on the UI suit most installations, you may want to adapt the
parameters for your specific deployment.

The heartbeat interval is
typically used to keep connections open through firewalls. Do not set the
Database Connection Heartbeat Interval value to zero without contacting Cisco
support.

(Optional)Check
Archive all room joins and exits if you want to log
all instances of users joining and leaving a room. This is a cluster-wide
setting that applies to all persistent chat rooms.

Step 4

(Optional)Check
Archive all room messages if you want to archive all
the messages that are sent in the room. This is a cluster-wide setting that
applies to all persistent chat rooms.

Step 5

(Optional) Check
Allow
only group chat system administrators to create persistent chat
rooms if you want to ensure that persistent chat rooms are created
only by group chat system administrators. This is a cluster-wide setting that
applies to all persistent chat rooms.
To configure
group chat system administrators, choose
Messaging > Group chat system
administrators.

Step 6

Enter the
maximum number of persistent chat rooms that are allowed in the
Maximum number of persistent chat rooms allowed
field. The default value is set to 1500.

Note

You must
ensure there is sufficient space on the external database. Having a large
number of chat rooms impacts resources on the external database.

Step 7

Enter the
number of connections to the database that you to want to use for processing
requests in the
Number of connections to the database field. The
default is set to 5. This is a cluster-wide setting that applies to all
connections between chat nodes and associated databases.

Step 8

Enter the
number of seconds after which the database connection should refresh in the
Database connection heartbeat interval (seconds)
field. The default is set to 300. This is a cluster-wide setting that applies
to all connections between chat nodes and associated databases.

Step 9

Enter the
number of minutes after which the chat room should time out in the
Timeout value for persistent chat rooms (minutes)
field. The default is set to 0. The timeout is used to check whether a chat
room is idle and empty. If the room is found to be idle and empty, the room is
closed. With the default value set to 0, the idle check is disabled.

Step 10

Choose from
the list of preconfigured external databases and assign the appropriate
database to the chat node.

If you
turn on the
Archive all room joins and exits setting, Cisco
recommends that you monitor the performance of each external database that is
used for persistent chat. Expect an increased load on the database servers.

If you
turn on the
Archive all room messages setting, Cisco recommends
that you monitor the performance of each external database that is used for
persistent chat. Expect an increased load on the database servers.

If you
enable persistent chat rooms but do not establish the correct connection with
the external database, the chat node will fail. Under these circumstances, you
will lose the functionality of all chat rooms, both temporary and persistent.
If a chat node establishes a connection (even if other chat nodes fail), it
will still start.

Click the
hyperlink if you need to edit the Cisco Unified Communications Manager IM and
Presence Service node details in the
Cluster Topology Details window.

After you
have enabled persistent chat, if you subsequently want to update any of the
persistent chat settings, only the following non-dynamic settings require a
Cisco XCP Text Conference Manager restart:

Number of connections to
the database

Database connection
heartbeart interval (seconds)

Set Number of Chat
Rooms

Use room settings to limit
the number of rooms that users can create. Limiting the number of chat rooms
will help the performance of the system and allow it to scale. Limiting the
number of rooms can also help mitigate any possible service-level attacks.

Procedure

Step 1

To change the
maximum number of chat rooms that are allowed, enter a value in the field for
maximum number of rooms allowed. The default is set
to 16500.

Step 2

Click
Save.

Configure Member
Settings

Member settings
allow system-level control over the membership in chat rooms. Such a control is
useful for users to mitigate service-level attacks that can be prevented by
administrative actions such as banning. Configure the member settings as
required.

Procedure

Step 1

Check
Rooms
are for members only by default if you want rooms to be created as
members-only rooms by default. Members-only rooms are accessible only by users
on a white list configured by the room owner or administrator. The checkbox is
unchecked by default.

Note

The white
list contains the list of members who are allowed in the room. It is created by
the owner or administrator of the members-only room.

Step 2

Check
Room
owners can change whether or not rooms are for members only if you
want to configure the room so that room owners are allowed to change whether or
not rooms are for members only. The check box is checked by default.

Note

A room owner
is the user who creates the room or a user who has been designated by the room
creator or owner as someone with owner status (if allowed). A room owner is
allowed to change the room configuration and destroy the room, in addition to
all other administrator abilities.

Step 3

Check
Only
moderators can invite people to members-only rooms if you want to
configure the room so that only moderators are allowed to invite users to the
room. If this check box is unchecked, members can invite other users to join
the room. The check box is checked by default.

Step 4

Check
Room
owners can change whether or not only moderators can invite people to
members-only rooms if you want to configure the room so that room
owners can allow members to invite other users to the room. The check box is
checked by default.

Step 5

Check
Users
can add themselves to rooms as members if you want to configure the
room so that any user can request to join the room at any time. If this check
box is checked, the room has an open membership. The check box is unchecked by
default.

Step 6

Check
Room
owners can change whether users can add themselves to rooms as
members if you want to configure the room so that room owners have
the ability to change the setting that is listed in Step 5 at any time. The
check box is unchecked by default.

Step 7

Click
Save.

Configure
Availability Settings

Availability
settings determine the visibility of a user within a room.

Procedure

Step 1

Check
Members and administrators who are not in a room are still
visible in the room if you want to keep users on the room roster
even if they are currently offline. The check box is checked by default.

Step 2

Check
Room
owners can change whether members and administrators who are not in a room are
still visible in the room if you want to allow room owners the
ability to change the visibility of a member or administrator. The check box is
checked by default.

Step 3

Check
Rooms
are backwards-compatible with older clients if you want the service
to function well with older Group Chat 1.0 clients. The check box is unchecked
by default.

Step 4

Check
Room
owners can change whether rooms are backwards-compatible with older
clients if you want to allow room owners the ability to control
backward compatibility of the chat rooms. The check box is unchecked by
default.

Step 5

Check
Rooms
are anonymous by default if you want the room to display the user
nickname but keep the Jabber ID private. The check box is unchecked by default.

Step 6

Check
Room
owners can change whether or not rooms are anonymous if you want to
allow room owners to control the anonymity level of the user Jabber ID. The
check box is unchecked by default.

Step 7

Click
Save.

Configure Invite
Settings

Invite settings
determine who can invite users to a room based on the user's role. Roles exist
in a moderator-to-visitor hierarchy so, for instance, a participant can do
anything a visitor can do, and a moderator can do anything a participant can
do.

Procedure

Step 1

From the
drop-down list for
Lowest
participation level a user can have to invite others to the room,
choose one:

Visitor allows
visitors, participants, and moderators the ability to invite other users to the
room.

Participant
allows participants and moderators the ability to invite other users to the
room. This is the default setting.

Moderator
allows only moderators the ability to invite other users to the room.

Step 2

Check
Room
owners can change the lowest participation level a user can have to invite
others to the room to allow room owners to change the settings for
the lowest participation level that is allowed to send invitations. The check
box is unchecked by default.

Step 3

Click
Save.

Configure
Occupancy Settings

Procedure

Step 1

To change the
system maximum number of users that are allowed in a room, enter a value in the
field for
How
many users can be in a room at one time. The default value is set
to 1000.

Note

The total
number of users in a room should not exceed the value that you set. The total
number of users in a room includes both normal users and hidden users.

Step 2

To change the
number of hidden users that are allowed in a room, enter a value in the field
for
How
many hidden users can be in a room at one time. Hidden users are
not visible to others, cannot send a message to the room, and do not send
presence updates. Hidden users can see all messages in the room and receive
presence updates from others. The default value is 1000.

Step 3

To change the
default maximum number of users that are allowed in a room, enter a value in
the field for
Default maximum occupancy for a room. The default
value is set to 50 and cannot be any higher than the value that is set in Step
1.

Step 4

Check
Room
owners can change default maximum occupancy for a room if you want
to allow room owners to change the default maximum room occupancy. The check
box is checked by default.

Step 5

Click
Save.

Configure Chat
Message Settings

Use Chat Message
settings to give privileges to users based on their role. For the most part,
roles exist in a visitor-to-moderator hierarchy. For example, a participant can
do anything a visitor can do, and a moderator can do anything a participant can
do.

Procedure

Step 1

From the
drop-down list for
Lowest
participation level a user can have to send a private message from within the
room, choose one:

Visitor allows
visitors, participants, and moderators to send a private message to other users
in the room. This is the default setting.

Participant
allows participants and moderators to send a private message to other users in
the room.

Moderator
allows only moderators to send a private message to other users in the room.

Step 2

Check
Room
owners can change the lowest participation level a user can have to send a
private message from within the room if you want to allow room
owners to change the minimum participation level for private messages. The
check box is checked by default.

Step 3

From the
drop-down list for
Lowest
participation level a user can have to change a room's subject,
choose one:

Participant
allows participants and moderators to change the room's subject. This is the
default setting.

Moderator
allows only moderators to change the room's subject.

Visitors are
not permitted to change the room subject.

Step 4

Check
Room
owners can change the lowest participation level a user can have to change a
room's subject if you want to allow room owners to change the
minimum participation level for updating a room's subject. The check box is
checked by default.

Step 5

Check
Remove
all XHTML formatting from messages if you want to remove all
Extensible Hypertext Markup Language (XHTML) from messages. The check box is
unchecked by default.

Step 6

Check
Room
owners can change XHTML formatting setting if you want to allow
room owners to change the XHTML formatting setting. The check box is unchecked
by default.

Step 7

Click
Save.

Configure
Moderated Room Settings

Moderated rooms
provide the ability for moderators to grant and revoke the voice privilege
within a room (in the context of Group Chat, voice refers to the ability to
send chat messages to the room). Visitors cannot send instant messages in
moderated rooms.

Procedure

Step 1

Check
Rooms
are moderated by default if you want to enforce the role of
moderator in a room. The check box is unchecked by default.

Step 2

Check
Room
owners can change whether rooms are moderated by default if you
want to allow room owners the ability to change whether rooms are moderated.
The check box is checked by default.

Step 3

Click
Save.

Configure History
Settings

Use History
settings to set the default and maximum values of messages that are retrieved
and displayed in the rooms, and to control the number of messages that can be
retrieved through a history query. When a user joins a room, the user is sent
the message history of the room. History settings determine the number of
previous messages that the user receives.

Procedure

Step 1

To change the
maximum number of messages that users can retrieve from the archive, enter a
value in the field for
Maximum number of messages that can be retrieved from the
archive. The default value is set to 100. It serves as a limit for
the next setting.

Step 2

To change the
number of previous messages displayed when a user joins a chat room, enter a
value in the field for
Number
of messages in chat history displayed by default. The default value
is set to 15 and cannot be any higher than the value that is set in Step 1.

Step 3

Check
Room
owners can change the number of messages displayed in chat history
if you want to allow room owners to change the number of previous messages
displayed when a user joins a chat room. The check box is unchecked by default.

Step 4

Click
Save.

IM and Presence
Service Group Chat System Administration

This section
describes how to configure group chat system administration.

Note

Jabber does not
currently support the persistent chat feature. The availability of the
functionality depends on client implementation.

Group chat system
administrators can do the following:

Configure a room

Join a
password-protected room without supplying the password

Change a room's
subject

Join any room
(including members-only rooms)

Moderate a room

Join a room
when the maximum occupancy is reached

Destroy a room

Browse a room
for the list of participants

Query a room and
its items

Remain in a
room if the room changes to be members-only, or if their affiliation changes to
"none" in a members-only room

Change the
affiliation of other users in a room

Invite other
users to a members-only room (even when members invite is not allowed).

Multiple
domain support. IM addresses do not need to use a single
IM and Presence
Service domain.

Alignment
with the user's email address. The
Cisco Unified Communications Manager Directory URI can
be configured to align with a user's email address to provide a consistent
identity for email, IM, voice and video communications.

Alignment
with Microsoft SIP URI. The
Cisco Unified Communications Manager Directory URI can
be configured to align with the Microsoft SIP URI to ensure that the
user's identity is maintained when migrating from Microsoft OCS/Lync
to
IM and Presence
Service.

You set the Directory URI using Cisco Unified CM IM and Presence
Administration GUI in one of two ways:

Synchronize
the Directory URI from the LDAP directory source.
If you add an
LDAP directory source in
Cisco Unified Communications Manager, you can set a
value for the Directory URI.
Cisco Unified Communications Manager then populates
the Directory URI when you synchronize user data from the directory source.

Note

If LDAP
Directory Sync is enabled in
Cisco Unified Communications Manager, you can map the
Directory URI to the email address (mailid) or the Microsoft OCS/Lync SIP URI
(msRTCSIP-PrimaryUserAddress).

Manually
specify the Directory URI value in
Cisco Unified Communications Manager.
If you do not
add an LDAP directory source in
Cisco Unified Communications Manager, you can manually
enter the Directory URI as a free-form URI.

See the
Cisco Unified
Communications Manager Administration Guide for more information about
setting up the LDAP directory for Directory URI.

IM and Presence Service supports IM addressing across multiple IM address domains and automatically lists all domains in the system. Use the Cisco Unified CM IM and Presence Administration GUI to manually add, update, and delete local administrator-managed domains, as well as view all local and system managed domains.

For more
information, see the
Deployment
Guide for IM and Presence
Service on
Cisco Unified Communications Manager.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk Administration
Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT Considerations

The following
alerts were added:

DuplicateDirectoryURI

InvalidDirectoryURI

DuplicateUserid

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

IM and Presence Service
Considerations

Use the Cisco
Unified CM IM and Presence Administration GUI to set up IM addressing and
multiple domain support for your deployment:

Directory
URI IM addressing scheme support and configuration

IM and Presence Service default domain changes

Multiple
domain setup and management

Multiple
domain support for Partitioned Intradomain Federation

End user
management and duplicate or invalid user entry troubleshooting

For details to
configure the IM addressing scheme, integrate and mange multiple domains, and
manage end users for your deployment, see the following guides:

Deployment Guide for
IM and Presence
Service on
Cisco Unified
Communications Manager

Interdomain Federation
for
IM and Presence
Service on Cisco Unified Communications Manager

Partitioned Intradomain
Federation for IM and Presence Service on Cisco Unified Communications
Manager

Note

If the
Directory URI IM address scheme is used anywhere in the deployment, your client
software must support Directory URI.

For multiple
domain support with intercluster deployments,
IM and Presence
Service clusters and client devices are no longer required to have
matching DNS domains. As well, the
IM and Presence
Service default domain no longer has to match the DNS domain. For
more information, see
Interdomain
Federation for
IM and Presence
Service on
Cisco Unified Communications Manager.

For information
about setting or changing the domain name for the
IM and Presence
Service node, see
Installing
the
Cisco Unified Communications Manager and
Changing
the Hostname and IP Address for
Cisco Unified Communications Manager and
IM and Presence
Service.

For multiple
domain support with partitioned intradomain federation, identical domains must
be configured on both the
IM and Presence
Service node and the supported Microsoft servers. For more
information, see
Partitioned
Intradomain Federation for
IM and Presence
Service on
Cisco Unified Communications Manager.

Procedure Changes

See the related
IM and Presence
Service guides for all the latest procedures.

UserID@Default_Domain IM Address Interactions and Restrictions

The following restrictions apply to the UserID@Default_Domain IM address scheme:

All IM addresses are part of the IM and Presence default domain, therefore, multiple domains are not supported.

The IM address scheme must be
consistent across all IM and Presence Service
clusters.

The default domain value must be consistent across all clusters.

If UserID is mapped to an LDAP field on Cisco Unified Communications Manager, that LDAP mapping must be consistent across all clusters.

Directory URI IM
Address Interactions and Restrictions

To support
multiple domain configurations, you must set Directory URI as the IM address
scheme for
IM and Presence Service.

Caution

If you
configure the node to use Directory URI as the IM address scheme, Cisco
recommends that you deploy only clients that support Directory URI. Any client
that does not support Directory URI will not work if the Directory URI IM
address scheme is enabled. Cisco recommends that you use the UserID@Default_Domain IM
address scheme and not the Directory URI IM address scheme if you have any
deployed clients that do not support Directory URI.

Observe the
following restrictions and interactions when using the Directory URI IM address
scheme:

All deployed
clients must support Directory URI as the IM address and use EDI-based
directory integration.

UDS-based
directory integration is not supported.

The IM address
scheme must be consistent across all
IM and Presence
Service clusters.

All clusters
must be running a version of
Cisco Unified Communications Manager that supports the
Directory URI addressing scheme.

If LDAP Sync
is disabled, you can set the Directory URI as a free-form URI. If LDAP
Directory Sync is enabled, you can map the Directory URI to the email address
(mailid) or the Microsoft OCS/Lync SIP URI (msRTCSIP-PrimaryUserAddress).

The Directory
URI IM address settings are global and apply to all users in the cluster. You
cannot set a different Directory URI IM address for individual users in the
cluster.

Your Cisco Jabber client must support Directory URI. See the documentation that came with
your Cisco Jabber client to determine compatibility.

The Cisco
Jabber client must be configured to align with the IM address scheme and the
Directory URI configuration on
IM and Presence
Service. By default, Cisco Jabber assumes the default IM address
scheme
UserID@Default_Domain. If Directory URI is used, then additional
configuration is required on the Cisco Jabber client to ensure that directory
searches align with the Directory URI value.
For example,
if the IM address scheme is Directory URI and that is mapped to mail in Active
Directory, then Jabber for windows directory searches against Active Directory
must be configured to ensure that the mail field is used as the IM address when
adding a contact. See the installation and configuration guide for your verison
of Cisco Jabber for Windows for details.

Note

To configure
the Directory URI IM address scheme for the Cisco Jabber client, you must
manually edit a configuration file in xml format. The xml configuration file
must be valid before you upload the file to the TFTP server. The Cisco Jabber
client ignores invalid configuration files.

CLI changes

utils users
validate

This command
checks user records across all nodes and clusters in the deployment to identify
duplicate or invalid userid or directory URI values.

utils users validate
{ all | userid | uri }

Syntax Description

Parameters

Description

all

Validate
the userid and directory URI values for all users in the nodes and clusters.

userid

Validate
the userid value for all users in the nodes and clusters.

uri

Validate
the directory URI value for all users in the nodes and clusters.

Command Modes

Administrator (admin:)

Requirements

Command privilege
level: 1

Allowed during
upgrade: No

Applies to:
IM and Presence Service on
Unified Communications Manager

IM and Presence
Service Oracle Database Support

You can configure
Oracle 11G, 10G and 9G as an external database to store information
synchronized from the Cisco Unified Communications Manager IM and Presence
Service.

Install Oracle
Database

Read the
security recommendations for the Oracle database in your Oracle documentation.

IM and Presence Service supports Oracle 11G,
10G and 9G.

Note

Cisco
recommends that an Oracle DBA install the Oracle server.

In compliance
with XMPP specifications, the
IM and Presence Service server uses UTF8
character encoding. This allows the server to operate using many languages
simultaneously and to display special language characters correctly in the
client interface. If you want to use Oracle with the server, you must configure
it to support UTF8.

To install the
Oracle database, refer to your Oracle documentation.

To create
tablespace and a database user, connect to the Oracle database as sysdba:

sqlplus
/ as sysdba

Procedure

Step 1

Create
tablespace.

Note

The DATAFILE keyword of the
CREATE TABLESPACE command tells Oracle where to put the
tablespace's datafile.

Prerequisite
Configuration Tasks for Oracle

Before
you configure IM compliance, make sure that you have performed the following
tasks:

Install the
IM and
Presence servers as described in the
Installing
Cisco Unified Communications Manager.

Configure the
IM and
Presence servers as described in the
Deployment
Guide for
IM and
Presence Service on Cisco Unified Communications Manager
.

Set up the external
database as described in the
Database
Setup for
IM and
Presence Service on Cisco Unified Communications Manager
.

Support for Oracle

In compliance with XMPP
specifications, the
IM and
Presence server uses UTF8 character encoding. This allows the server
to operate using many languages simultaneously and to display special language
characters correctly in the client interface. If you want to use Oracle with
the server, you must configure it to support UTF8.

The value of the
NLS_LENGTH_SEMANTIC
parameter should be set to
BYTE.

To determine the tablespace
available for your Oracle database, execute the following query as sysdba:
SELECT DEFAULT_TABLESPACE
FROM DBA_USERS WHERE USERNAME = 'UPPER_CASE_USERNAME';

Set Up Oracle
Database Entry

Perform
this configuration on the publisher node of your
IM and
Presence Service cluster.

Before You Begin

Install and configure the
external database.

Obtain the hostname or IP
address of the external database.

Retrieve the tablespace
value. To determine the tablespace available for your Oracle database, execute
the following query as sysdba:
SELECT DEFAULT_TABLESPACE
FROM DBA_USERS WHERE USERNAME = 'UPPER_CASE_USERNAME';

Enter the name
of the database that you defined at external database installation, for example
"tcmadb."

Step 4

Enter the
tablespace value.

Step 5

Enter the
username for the database user (owner) that you defined at external database
installation, for example
"tcuser."

Step 6

Enter and
confirm the password for the database user, for example
"mypassword."

Step 7

Enter the
hostname or IP address for the external database.

Step 8

Enter a port
number for the external database.

The default
port number for Oracle (1521) will be pre-populated in the Port Number field.
You can choose to enter a different port number if required.

Step 9

Select
Save.

IM and Presence
Service Support for Microsoft Lync Server 2013

IM
and Presence Service supports Microsoft Lync Server 2013. When
entering the Microsoft server type during federation configuration, you can
enter either Microsoft Lync Server 2010 or 2013. For more information about
Lync server support, see the
Partitioned
Intradomain Federation for
IM and Presence
Service on
Cisco Unified Communications Manager and
Interdomain
Federation for
IM and Presence
Service on
Cisco Unified Communications Manager.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk Administration
Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Feature Group
Template Support for the User Synced From Directory

Cisco Unified Communications Manager, Release 10.0(1)
provides
Feature
Group Template and
Access
Control Groups support for new users. The administrator can select
both the
Feature
Group Template and
Access
Control Groups and integrate them with the new users that are
synchronized from the LDAP directory.

Cisco Unified
Communications Manager Administration Considerations

LDAP directory
settings

The following
table describes the LDAP directory settings.

Table 15 LDAP Directory
Settings

Field

Description

Work
Number

(drop-down list box)

For
these fields, the
Cisco Unified
Communications Manager data in the field specified at left gets
synchronized with the LDAP user data in the field specified at right.

For the
LDAP User field, choose one of the following values:

telephoneNumber

ipPhone

Title

title

For
these fields, the
Cisco Unified
Communications Manager data in the field specified at left gets
synchronized with the LDAP user data in the field specified at right.

Mobile
Number

mobile

For
these fields, the
Cisco Unified
Communications Manager data in the field specified at left gets
synchronized with the LDAP user data in the field specified at right.

Home
Number

homePhone

For
these fields, the
Cisco Unified
Communications Manager data in the field specified at left gets
synchronized with the LDAP user data in the field specified at right.

Pager
Number

pager

For
these fields, the
Cisco Unified
Communications Manager data in the field specified at left gets
synchronized with the LDAP user data in the field specified at right.

Group
Information

Access
Control Groups

Use this
option to manage the Access Control Group to configure different levels of
access for new users that were synchronized from the LDAP directory.

Click
the
Add to Access Control Group button to open the Find
and List Access Control Groups window. From the list, select one or more Access
Control Groups for a user. Click the
Add Selected button. The Find and List Access
Control Groups window closes, and the Update Users Configuration window now
shows the selected groups in the list box.

To
delete an existing Access Control Group, select the relevant Access Control
Group from the list box. Click the
Remove from Access Control button to complete the
process.

To add
a new Access Control Group to the Find and List Access Control Groups window,
use the following menu path: User
Management > User Settings > Access Control
Group

Feature
Group Template

From the
drop-down list box, select the Feature Group template to be associated with the
new users that are synchronized from the LDAP directory.

To
create a Feature Group template that includes features such as mobility and IM
and Presence, use the following menu path:User
Management > User/Phone Add > Feature Group
Template

If you
do not select a feature group template, a warning message displays as mentioned
below:

Warning

If
no template is selected, the new line features below will not be active.

If you
select a custom feature group template with no user profile, a warning message
displays as mentioned below:

Warning

The
selected Feature Group Template does not have a Universal Line Template
configured. The new line features below will not be active.

End user settings

The following
table describes the end user settings.

Table 16 End User
Settings

Field

Description

Title

Enter
the end user title.

Work
Number

Enter
the end user work number. You may use the following special characters: (, ),
and -.

Mobile
Number

Enter
the end user mobile number. You may use the following special characters: (, ),
and -.

Home
Number

Enter
the end user home number. You may use the following special characters: (, ),
and -.

Pager
Number

Enter
the end user pager number. You may use the following special characters: (, ),
and -.

Bulk
Administration Considerations

User Update Settings

The
following table provides descriptions for all possible fields when you update
users with the Query option.

Table 17 Field
Descriptions for Update Users

Field

Description

User
Information

Manager
User ID

Enter
manager user ID, up to 128 characters, for the user of this phone.

Department

Enter
the department number, up to 64 characters, for the user of this phone.

Associated PC

This
field, which is required for
Cisco SoftPhone and
Cisco Unified Communications Manager Attendant Console
users, displays after you add the user.

User
Locale

Choose
the language and country set that you want to associate with this user from the
drop-down list. Your choice determines which cultural-dependent attributes
exist for this user and which language displays in the Cisco Unified
Communications Manager user windows and phones.

Digest
Credentials

When you
configure digest authentication for phones that are running,
Cisco Unified Communications Manager challenges the
identity of the phone every time the phone sends a SIP request to
Cisco Unified Communications Manager. The digest
credentials that you enter in this field get associated with the phone when you
choose a digest user in the Phone Configuration window.

Enter a
string of up to 128 alphanumeric characters.

For more
information on digest authentication, see the
Cisco
Unified Communications Manager Security Guide.

Confirm
Digest Credentials

To
confirm that you entered the digest credentials correctly, reenter the
credentials in this field.

Service
Setting

UC
Service Profile

Select a
UC service profile from the drop-down list box to associate with end users.

Note

Use
the
User
Management > User Settings > Service Profile
menu to set up service profiles
for end users.

Include
meeting information in Presence

Check
this check box to enable the end user to include meeting and calendar
information in
IM and Presence Service.

The end
user must be on the home cluster and have IM and Presence enabled. Also ensure
that an Exchange Presence Gateway is configured on the
Cisco Unified Communications Manager IM and Presence
Service server.

Include
Meeting Information in Presence (Requires Exchange Presence Gateway to be
configured on CUCM IM and Presence server)

Check
this check box to create a sync between CUCM IM and Presence server so that it
can include the meeting information under the Presence feature.

Note

You
can only access this field if Home Cluster and Enable User for Unified CM IM
and Presence is enabled.

Extension Mobility

BLF
Presence Group

From the
drop-down list, choose the BLF presence group that watches the status of the
directory number, the presence entity.

For
information on the BLF Presence feature, see the
Cisco
Unified Communications Manager Features and Services Guide.

The
SUBSCRIBE Calling Search Space determines how Cisco Unified
Communications Manager routes the Presence subscription requests that
come from the end user. Use the
Call
Routing > Class Control > Calling Search
Space menu to configure a calling search space
specifically for this purpose .

For
information on how to configure a calling search space, see the
Cisco
Unified Communications Manager Administration Guide.

Allow
Control of Device from CTI

Check
this check box to allow CTI to control and monitor this device.

If the
associated directory number specifies a shared line, the check box should be
checked as long as at least one associated device specifies a combination of
device type and protocol that CTI supports.

Enable
Extension Mobility Cross Cluster

Check
this check box to enable the Extension Mobility Cross Cluster CSS setting that
gets used as the device CSS of the remote phone when the user selects this
device profile during EMCC login.

Mobility
Information

Enable
Mobility

Check
this check box to activate Mobile Connect, which allows the user to manage
calls by using a single phone number and to pick up and manage calls on the
desk phone and mobile phone.

RTMT
Considerations

Security
Considerations

Serviceability
Considerations

You can
update a group of user records in the
Cisco Unified Communications
Manager directory.

Before You Begin

You must
have a .csv data file with updated user information.

Procedure

Step 1

Choose
Bulk
Administration > Users > Update Users > Custom
File.

The
User
Update Configuration window appears.

Step 2

From the
File
Name drop-down list box, choose the .csv data file that you created
for this bulk transaction.

Note

Click
View
File to view the uploaded .csv data file.

Click
View Sample
File to view a sample .csv data file.

Step 3

From the
User
Template Name drop-down list box, choose the user template that you
created for this bulk transaction.

Step 4

In the
Value
for fields to be ignored field, enter the symbol that you want to
tell Unified CM Bulk Administration Tool to retain the value that was
previously stored in the DC directory.

Note

The value that
you enter in the .csv file for updating users overrides the values that are
provided in the user template.

Step 5

In the
Job
Information area, enter the Job description.

Step 6

Choose a method
to update user records. Do one of the following:

Select
Run
Immediately to update user records immediately.

Select
Run
Later to insert the user records at a later time.

Step 7

To create a job
for updating the user records, click
Submit.
To schedule or
activate this job, use the Job Scheduler option in the Bulk Administration main
menu.

IPMA and Softkey
Template Support

For
Unified Communications Manager, Release 10.0(1) and
later, the IP Manager Assistant feature is managed by application on the phone.
This is not modified by user information and updates to user settings.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

A list of phones
supporting this feature is available in the
Cisco Unified
Communications Manager Assistant User Guide.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

IPv6 Enterprise Parameters

Cisco Unified Communications Manager Administration Considerations

The following new IPV6 enterprise parameters have been added (System > Enterprise Parameter Configuration):

Allow Duplicate Address Detection

Accept Redirect Messages

Reply Multicast Echo Request

Cisco Unified Communications Manager (Unified Communications Manager) also provides an option to configure these parameters in the Common Device Configuration window.

The following fields have been added to the Common Device Configuration window (Device > Device Settings > Common Device Configuration):

If you set these parameters as 'Default', the configuration file of IP Phones will pick up the values of these parameters from the enterprise parameters. If you set these parameters as 'On' or 'Off', the configuration file will consider these values.

Serviceability
Considerations

IPv6-Only and Dual
Stack SIP Endpoints

Cisco Unified
Communications Manager Administration Considerations

IP Phones or other
endpoints that use Session Initiation Protocol (SIP) can register with
Cisco Unified Communications Manager (Unified Communications Manager) using an IPv6 address.
After these phones have registered with
Unified Communications Manager, they can operate in
IPv6 mode, IPv4 mode, or in dual stack mode. Administrators can specify the
order of address preference for these phones by using the IP Addressing Mode
Preference for Signaling setting, which is located on the Common Device
Configuration panel.

This feature also
adds support for all media types, including audio and video, for calls
originating from and terminating on IPv6 SIP line devices located in the same
cluster, or in different clusters connected by SIP trunks.

You can use the
ANAT Enabled checkbox, which is selected by default, to enable or disable this
feature. The checkbox was previously located on the SIP Profile panel, and is
now located on the SIP Trunks configuration panel.

The following
phones support IPv6 and dual stack addressing:

Cisco Unified
SIP Phone 3905

Cisco Unified
IP Phone 7821

Cisco Unified
IP Phone 7841

Cisco Unified
IP Phone 7845

Cisco Unified
IP Phone 7861

Cisco Unified
IP Phone 8961

Cisco Unified
IP Phone 9951

Cisco Unified
IP Phone 9971

The following SIP
Telepresence endpoints support IPv6 and dual stack addressing:

C-series (C90,
C60, C40, C20)

Profile-series

SX-series
(SX20)

MX-series
(MX200, MX300)

EX-series
(EX60, EX90)

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

iX Channel MTP
Transparency

iX channel over
MTP/RSVP provides a simple, reliable and secure channel for multiplexing
multiple application layer protocols. The transport used for iX channel is User
Datagram Protocol (UDP). To provide a reliable channel, iX channel over
MTP/RSVP uses UDP Based Data Transfer Protocol (UDT) over UDP. To provide
security, Transportation Layer Security (TLS) is run over the reliable
transport provided by UDT. iX channel over MTP/RSVP provides a multiplexer,
which uses a simple type-length-value scheme that enables the channel to
support application layer protocols of many types, including XML and binary
protocols.

iX channel over
MTP/RSVP can be negotiated and set up using the Session Description Protocol
(SDP) and the Offer/Answer model. An iX channel extends SDP to support new
attribute mapping for the protocols to be multiplexed.

To support iX
channels in MTP cases,
Unified
Communications Manager must be configured to invoke MTPs that are
allocated from the IOS router, which must be running version 15.2T or above.

Note

Unified Communications
Manager will only support iX channel negotiation when a video channel
has been established.

For RSVP and E2E
RSVP calls,
Unified
Communications Manager supports iX channel negotiation and allows iX
channels to pass through the RSVP agents.
Unified
Communications Manager does not need to reserve any bandwidth for iX
channels in RSVP agents. RSVP will continue to reserve the bandwidth for audio
and the primary video channel only.

You can only
transfer a maximum of 20 languages as user-locale prompts in
Cisco Unified Communications Manager.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Mobility Billing

The following call detail record (CDR) fields have been added for the 10.0(1) release for calls that invoke a mobility feature. If the call does not invoke a mobility feature, these fields remain empty:

MobileCallingPartyNumber

FinalMobileCalledPartyNumber

OrigMobileDeviceName

DestMobileDeviceName

OrigMobileCallDuration

DestMobileCallDuration

MobileCallType

MobileCallType Values

The MobileCallType CDR field has been added to identify the mobility feature that is invoked.

The following table displays the field values for the MobilityCallType CDR field. If a single call invokes more than one mobility feature, the value of the MobileCallType field will represent the integer values added together. For example, if a call uses the Mobile Connect feature and then invokes Hand-Out, the mobile call type will be 132 (8 + 128).

Table 18 MobilityCallType CDR Field Values

Mobility Feature

MobileCallType Value

Nonmobility call

0

Dial via Office Reverse Callback

1

Dial via Office Forward

2

Reroute Remote Destination Call to Enterprise Network

4

Mobile Connect

8

Interactive Voice Response

16

Enterprise Feature Access

32

Hand-In

64

Hand-Out

128

Redial

256

Least Cost Routing with Dial via Office Reverse Callback

512

Least Cost Routing with Dial via Office Forward

130

Send Call to Mobile

2048

Session Handoff

4096

Last Redirect Reason

In legacy deployments prior to 10.0(1), CAR uses the LastRedirectReason field to identify the mobility call type.

The following table shows the Mobility values for LastRedirectReason.

Table 19 Mobility Values for the LastRedirectReason Field

Mobility Feature

LastRedirectReason Value

Hand-In

303

Hand-Out

319

Mobile Connect

335

Redial

351

Interactive Voice Response

399

Dial via Office Reverse Callback

401

Enterprise Feature Access

402

Session Handoff

403

Least Cost Routing with Dial via Office Forward

404

Least Cost Routing with Dial via Office Reverse Callback

405

Send Call to Mobile

415

Reroute Remote Destination Call to Enterprise Network

783

Cisco Unified Communications Manager Administration Considerations

No changes.

Bulk Administration Considerations

No changes.

CDR/CAR Considerations

For examples of CDRs that are produced for calls that invoke specific mobility features, see the "CDR Examples" chapter of the Cisco Unified Communications Manager Call Detail Records Administration Guide.

IP Phones Considerations

No changes.

RTMT Considerations

No changes.

Security Considerations

No changes.

Serviceability Considerations

No changes.

Multiple Codecs in
SDP Answer

This option
applies when incoming SIP signals do not indicate support for multiple codec
negotiation and
Cisco
Unified Communications Manager can finalize the negotiated codec.

When this check
box is checked, the endpoint behind the trunk is capable of handling multiple
codecs in the answer SDP.

For example, an
endpoint that supports multiple codec negotiation calls the SIP trunk and
Cisco
Unified Communications Manager sends a Delay Offer request to a
trunk. The endpoint behind the trunk returns all support codecs without the
Contact header to indicate the support of multiple codec negotiation.

In this case,
Cisco
Unified Communications Manager identifies the trunk as capable of
multiple codec negotiation and sends SIP response messages back to both
endpoints with multiple common codecs.

Cisco Unified
Communications Manager Administration Considerations

The following
check box has been added to the
SIP
Profile settings window:
Allow
Multiple Codecs in Answer SDP.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Non-SRTP Call Blocking

The Non-SRTP Call Blocking feature enables you to block unencrypted (non-SRTP) calls. The calls are blocked if either the called party or the calling party is not encrypted. A new service parameter Block Unencrypted Calls has been added to enable this feature. By default, this service parameter is set to False. You must set this service parameter to True for blocking non-SRTP calls. When this service parameter is set to True and any one or each of the endpoints is unencrypted, the calls are blocked and Reorder tone is played on the endpoints. A perform counter gets incremented for each blocked call. An alarm is also raised for all the blocked calls so that the administrator can take appropriate action.

Bulk Administration Considerations

No changes.

CDR/CAR Considerations

No changes.

IP Phones Considerations

No changes.

RTMT Considerations

No changes.

Security Considerations

No changes.

Serviceability Considerations

No changes.

Plug and Play Feature

Cisco Unified Communications Manager Administration Considerations

Currently, all the IP phones are automatically registered with Cisco Unified Communications Manager. With the help of LDAP Sync and Bulk administration tool, a number of IP phones can be assigned to the users on the Cisco Unified Communications Manager. This feature provides an easy way to assign the auto-registered IP phones to the users.

For Auto-Registration settings, the options Partition and External Phone number Mask are replaced by Universal device Template and Universal Line Template.

Note

When you upgrade a previous release Cisco Unified Communications Manager to Release 10.0, the Cisco Unified Communications Manager will create a Universal Device Template and a Universal Line Template which will retain the previous configurations for Auto-Registration settings (Partition and External Phone Number Mask). After the upgrade, the Cisco Unified Communications Manager populates the Cisco Unified Communications Manager name for the Universal Device Template and a Universal Line Template and configures the same values for Auto-Registration settings.

For LDAP Directory settings, the following new fields are introduced.

Field

Description

Apply mask to synced telephone numbers to create a new line for inserted users

Check the check box to apply mask to the synced telephone number of the user.

Enter a mask value in the Mask text box. The Mask can contain one to twenty four characters including numbers (0-9), X, and x. It must include at least one x or X.

For example, if you set the mask as 11XX for the user with a telephone number 8889945, after the mask is applied, 1145 becomes the primary extension of the user.

Assign new line from the pool list if one was not created based on a synced LDAP telephone number

Check the check box to assign a new line from the DN pool list.

Next Candidate DN

Displays the next probable DN that will be assigned to the user.

The DN from the next DN pool is displayed only after all the DNs from the first DN pool are assigned.

Note

The Next Candidate DN displays only when you check the Assign new line from the pool list if one was not created based on a synced LDAP telephone number check box.

Add DN Pool

By default, only one DN pool is available. Click this option to add more DNs to the DN pool.

The DN Pool Start and DN Pool End values must conform to the following requirements:

Must be a number and can contain one to twenty characters

DN Pool End must be greater than DN Pool Start

DN Pool Start and DN Pool End must not be null

DN range must be less than 10,000,000

Enter the DN Pool Start and DN Pool End values in the text box. You can reorder the DN pool to prioritize the DNs that you want to assign.

If the length of the start and end DN pools are different, an error message displays: The DNs length must be identical.

You can create only three DN pools.

Bulk Administration Considerations

The Import/Export support is extended to Self-Provisioning, where the administrators can export Self-Service User ID that is generated when a user is assigned an IP phone. The BAT spreadsheet now includes the Self-Service User ID for User Data option.

CDR/CAR Considerations

No Changes.

IP Phones Considerations

All phone models that support Auto-Registration supports Self-Provisioning. The Self-Provisioning process for desk phones, such as Jabber and Zydeco phones are different, compared to Self-Provisioning for IP phones. Self-Provisioning is also supported on analog phones.

RTMT Considerations

No Changes.

Security Considerations

No Changes.

Serviceability Considerations

This feature introduces a new Self-Provisioning Interactive Voice Response (IVR). When you dial the CTI RP DN, that is configured on the Self-Provisioning page, from an extension of a user that uses the IVR service, the phone connects to the Self-Provisioning IVR application and prompts you to provide the Self-Service credentials. Based on the validation of the Self-Service credentials that you provide, the IVR service assigns the autoregistered IP phones to the users.

You can activate, deactivate, or restart the Self-Provisioning IVR service from the Serviceability page. By default, this service is deactivated. You can still configure Self-Provisioning on the Administration page even if the service is deactivated, but you cannot assign IP phones to users using the IVR service.

To enable Self-Provisioning IVR service, you must also enable Cisco CTI Manager service.

Remote
Lock/Wipe

Cisco Unified
Communications Manager Administration Considerations

In Unified
Communications Manager, some phones can be locked remotely. When a remote lock
is performed on a phone, the phone cannot be used until it is unlocked.

If a phone
supports the Remote Lock feature, a
Lock button
appears in the top right hand corner.

In Unified
Communications Manager, some phones can be wiped remotely. When a remote wipe
is performed on a phone, the operation resets the phone to its factory
settings. Everything previously stored on the phone is wiped out.

If a phone
supports the Remote Wipe feature, a
Wipe button
appears in the top right hand corner.

Unified
Communications Manager provides a specific search window for searching for
devices which have been remotely locked and/or remotely wiped.

Bulk
Administration Considerations

Wipe or Lock Phones Using QueryYou
can create a query to locate phones that you want to wipe and/or lock.

Wipe or Lock Phones Using Custom File
You can
create a custom file of phones that you want to wipe and/or lock using a text
editor.

Click
Lock.
If the phone
is not registered, a popup window displays to inform you that the phone will be
locked the next time it is registered. Click
Lock.
A
Device
Lock/Wipe Status section appears, with information about the most recent
request, whether it is pending, and the most recent acknowledgement.

Remotely Wipe a
Phone

Follow these steps
to remotely wipe a phone.

Caution

This operation
cannot be undone. You should only perform this operation when you are sure you
want to reset the phone to its factory settings.

Procedure

Step 1

Choose
Device > Phone.

The
Find
and List Phones window displays.

Step 2

Enter search
criteria and click
Find to locate a specific phone.

A list of
phones that match the search criteria displays.

Step 3

Choose the
phone for which you want to perform a remote wipe.

The
Phone
Configuration window displays.

Step 4

Click
Wipe.
If the phone
is not registered, a popup window displays to inform you that the phone will be
wiped the next time it is registered. Click
Wipe.
A
Device
Lock/Wipe Status section appears, with information about the most recent
request, whether it is pending, and the most recent acknowledgment.

Display Phone
Wipe/Lock Report

Unified Communications Manager provides a specific
search window for searching for devices which have been remotely locked and/or
remotely wiped. Follow these steps to search for a specific device or to list
all devices which have been remotely locked and/or remotely wiped.

Procedure

Step 1

Choose
Device > Phone.

The Find and
List Phones window displays. Records from an active (prior) query may also
display in the window.

Step 2

Select the
Phone Lock/Wipe Report from the Related Links drop-down list box in the upper,
right corner of the Find and List Phones window and click Go. The Find and List
Lock and Wipe Devices window displays.

Step 3

To find all
remotely locked or remotely wiped device records in the database, ensure that
the text box is empty; go to Step 4.

To filter or
search records

From the
first drop-down list box, select the device operation type(s) to search.

From the
second drop-down list box, select a search parameter.

From the
third drop-down list box, select a search pattern.

Specify
the appropriate search text, if applicable.

Note

To add
additional search criteria, click the + button. When you add criteria, the
system searches for a record that matches all criteria that you specify. To
remove criteria, click the – button to remove the last added criterion or click
the Clear Filter button to remove all added search criteria.

Step 4

Click
Find.

All matching
records display. You can change the number of items that display on each page
by choosing a different value from the Rows per Page drop-down list box.

Step 5

From the list
of records that display, click the link for the record that you want to view.

Note

To reverse
the sort order, click the up or down arrow, if available, in the list header.

The window
displays the item that you choose.

Wipe or Lock
Phones Using Query

Caution

The wipe
operation cannot be undone. You should only perform this operation when you are
sure you want to reset the phone to its factory settings.

In the
Update
Phones where drop-down list box, choose the type of custom file
that you have created from one of the following criteria:

Device Name

Directory Number

Description

Step 3

In the list of
custom files, choose the filename of the custom file for this update and then
click
Find.

Caution

If no
information is entered into the query text box, the system wipes or locks all
phones.

Step 4

Click one of
the following:

Lock—To lock the phones

Wipe—To wipe the phones

Wipe and Lock—To wipe and
lock the phones

Note

If a phone
does not support the functionality you have chosen, the transaction will fail
for that phone. It will also fail if the functionality has already been
requested for the phone.

Step 5

In the
Job
Information area, enter the Job description.

Step 6

Choose an
insert method. Do one of the following:

Click
Run Immediately to wipe or lock phones immediately.

Click
Run Later to wipe or lock phones at a later time.

Step 7

To create a
job for locking and/or wiping the phones, click
Submit.
To schedule
and/or
activate this job, use the
Job
Configuration window.

Removal of ASCII
Fields

Cisco Unified
Communications Manager Administration Considerations

The following
ASCII fields have been removed from the Unified Communications Manager
interface:

ASCII Label

ASCII Display
Name

ASCII Service
Name

Bulk
Administration Considerations

The following
ASCII fields have been removed from the Bulk Administration Tool (BAT)
interface:

ASCII Label

ASCII Line
Text Label

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Routing
Enhancements

Cisco Unified
Communications Manager Administration Considerations

Directory number
settings

The following
table describes the fields that are available in the Directory Number
Configuration window.

Table 20 Directory
Number Settings

Field

Description

Directory Number Information

Urgent
Priority

If the
dial plan contains overlapping patterns,
Cisco Unified Communications Manager does not route
the call to the device associated with the directory number until the
interdigit timer expires (even if the directory number is a better match for
the sequence of digits dialed as compared to the overlapping pattern). Check
this check box to interrupt interdigit timing when
Cisco Unified Communications Manager must route a call
immediately to the device associated with the directory number.

By
default, the Urgent Priority check box is unchecked.

Translation pattern
settings

The following
table describes the available fields in the Translation Pattern Configuration
window.

Table 21 Translation
Pattern Settings

Field

Description

Pattern
Definition

Use
Originator's Calling Search Space

To use
the originator's calling search space for routing a call, check the Use
Originator's Calling Search Space check box. When you check this check box, it
disables the Calling Search Space drop-down list box. When you save the page,
the Calling Search Space box is grayed out and set to <None>.

If the
originating device is a phone, the originator's calling search space results
from the device calling search space (configured on the Phone Configuration
window) and line calling search space (configured on the Directory Number
Configuration window).

Whenever
a translation pattern chain is encountered, for subsequent lookups Calling
Search Space is selected depending upon the value of this check box at current
translation pattern. If you check the Use Originator's Calling Search Space
check box at current translation pattern, then originator's Calling Search
Space is used and not the Calling Search Space for the previous lookup. If you
uncheck the Use Originator's Calling Search Space check box at current
translation pattern, then Calling Search Space configured at current
translation pattern is used.

Do Not
Wait For Interdigit Timeout On Subsequent Hops

When you
check this check box along with the Urgent Priority check box and the
translation pattern matches with a sequence of dialed digits (or whenever the
translation pattern is the only matching pattern),
Cisco Unified Communications Manager does not start
the interdigit timer after it matches any of the subsequent patterns.
Note:
Cisco Unified Communications Manager does not start
the interdigit timer even if subsequent patterns are of variable length or if
overlapping patterns exist for subsequent matches.

Whenever
you check the Do Not Wait For Interdigit Timeout On Subsequent Hops check box
that is associated with a translation pattern in a translation pattern chain,
Cisco Unified Communications Manager does not start
the interdigit timer after it matches any of the subsequent patterns.
Note:
Cisco Unified Communications Manager does not start
interdigit timer even if subsequent translation patterns in a chain have Do Not
Wait For Interdigit Timeout On Subsequent Hops unchecked.

Call Control Discovery
feature parameters

To
access the feature parameters that support the call control discovery feature,
choose
Call
Routing > Call Control Discovery > Feature
Configuration. For additional information, you can
click the question mark help in the Feature Configuration window.

The following
table describes the feature parameters for the call control discovery feature.

Table 22 Call Control
Discovery Feature Parameters

Feature
Parameter

Description

Set
Urgent Priority for Fixed-Length CCD Learned Patterns

This
parameter determines whether
Cisco Unified Communications Manager waits for
interdigit timer before routing the call to the destination that is associated
with the fixed-length learned pattern (when the fixed-length learned pattern is
a better match for the sequence of digits dialed as compared to the overlapping
route pattern configured). If the parameter is set to True,
Cisco Unified Communications Manager does not wait for
the interdigit timer before routing the call to the destination that is
associated with the fixed-length learned pattern. If the parameter is set to
False,
Cisco Unified Communications Manager waits for
interdigit timer before routing the call to the destination that is associated
with the fixed-length learned pattern. The default equals False.

Example:
Cisco Unified Communications Manager learns the
pattern +44987XXX for routing the calls to another
Cisco Unified Communications Manager and there is also
a route pattern configured as \+44! for routing the calls to the PSTN
destination. If this parameter is set to False and +44987127 is dialed,
Cisco Unified Communications Manager waits for
interdigit timer before routing the call to another
Cisco Unified Communications Manager (this interdigit
timer allows user to dial more digits after +44987127 to reach the PSTN
destination). If this parameter is set to True and +44987127 is dialed, then
Cisco Unified Communications Manager immediately
routes the call to another
Cisco Unified Communications Manager.

Set
Urgent Priority for Variable-Length CCD Learned Patterns

This
parameter determines whether
Cisco Unified Communications Manager waits for
interdigit timer before routing the call to the destination that is associated
with the variable-length learned pattern. If the parameter is set to True,
Cisco Unified Communications Manager does not wait for
interdigit timer before routing the call to the destination that is associated
with the variable-length learned pattern. If the parameter is set to False,
Cisco Unified Communications Manager waits for
interdigit timer before routing the call to the destination that is associated
with the variable-length learned pattern. The default equals False.

Example:
Cisco Unified Communications Manager has translation
pattern 9011.!# configured. This translation pattern strips predot digits and
the trailing # character and adds the prefix +55 to the dialed digits. Cisco Unified
Communications Manager also learns pattern \+55.! for routing the
calls to another
Cisco Unified Communications Manager. If this parameter is set to
False and 9011234567# (resultant digits = +55234567) is dialed,
Cisco Unified Communications Manager waits for
interdigit timer before routing the call to another
Cisco Unified Communications Manager. If this
parameter is set to True and 9011234567# (resultant digits = +55234567) is
dialed, then
Cisco Unified Communications Manager immediately
routes the call to another
Cisco Unified Communications Manager.

The
following table describes the settings for SIP trunks.

Table 23 SIP Trunk
Settings

Field

Description

Incoming
Called Party Settings

Clear
Prefix Settings

To
delete the prefix for unknown number type for the called party, click Clear
Prefix Settings.

Default
Prefix Settings

To enter
the default value for the Prefix field for unknown number type, click Default
Prefix Settings.

Unknown
Number

Configure the following settings to transform incoming called
party numbers that use Unknown for the Called Party Number Type.

Prefix—Cisco Unified Communications Manager applies the
prefix that you enter in this field to called numbers that use Unknown for the
Called Party Number Type. You can enter up to 16 characters, which include
digits, the international escape character (+), asterisk (*), or the pound sign
(#). You can enter the word, Default, instead of entering a prefix.

Tip

If
the word, Default, displays in the Prefix field, you cannot configure the Strip
Digits field. In this case,
Cisco Unified Communications Manager takes the
configuration for the Prefix and Strip Digits fields from the device pool that
is applied to the device. If the word, Default, displays in the Prefix field in
the Device Pool Configuration window,
Cisco Unified Communications Manager does not apply
any prefix or strip digit functionality.

Tip

To
configure the Strip Digits field, you must leave the Prefix field blank or
enter a valid configuration in the Prefix field. To configure the Strip Digits
fields in these windows, do not enter the word, Default, in the Prefix field.

Strip Digits—Enter the number of digits that you want
Cisco Unified Communications Manager to strip from the
called party number of Unknown type before it applies the prefixes.

Calling Search Space—This setting allows you to transform the
called party number of Unknown called party number type on the device. If you
choose None, no transformation occurs for the incoming called party number.
Make sure that the calling search space that you choose contains the called
party transformation pattern that you want to assign to this device.

Use
Device Pool CSS—Check this check box to use the calling search space for the
Unknown Number field that is configured in the device pool that is applied to
the device.

Connected Party Settings

Connected Party Transformation CSS

This
setting is applicable only for inbound calls. This setting allows you to
transform the connected party number on the device to display the connected
number in another format, such as a DID or E164 number.
Cisco Unified
Communications Manager includes the transformed number in the headers
of various SIP messages, including 200 OK and mid-call update/reinvite
messages. Make sure that the Connected Party Transformation CSS that you choose
contains the connected party transformation pattern that you want to assign to
this device.

Note

If you
configure the Connected Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the Calling
Party Transformation pattern used for Connected Party Transformation in a
non-null partition that is not used for routing.

Outbound
Calls

Called
Party Transformation CSS

This
settings allows you to send the transformed called party number in INVITE
message for outgoing calls made over SIP Trunk. Make sure that the Called Party
Transformation CSS that you choose contains the called party transformation
pattern that you want to assign to this device.

Note

If
you configure the Called Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the Called
Party Transformation CSS in a non-null partition that is not used for routing.

Calling
Party Transformation CSS

This
settings allows you to send the transformed calling party number in INVITE
message for outgoing calls made over SIP Trunk. Also when redirection occurs
for outbound calls, this CSS will be used to transform the connected number
that is sent from Cisco Unified Communications Manager side in outgoing
reINVITE / UPDATE messages.

Make
sure that the Calling Party Transformation CSS that you choose contains the
calling party transformation pattern that you want to assign to this device.

Tip

If you configure the Calling Party Transformation CSS as None,
the transformation does not match and does not get applied. Ensure that you
configure the Calling Party Transformation Pattern in a non-null partition that
is not used for routing.

This
setting is applicable only for inbound Calls. This setting allows you to
transform the connected party number that
Cisco Unified Communications Manager sends in another
format, such as a DID or E.164 number. This setting is applicable while sending
connected number for basic call as well as sending connected number after
inbound call is redirected.

Cisco Unified Communications Manager includes the
transformed number in the Connected Number Information Element (IE) of CONNECT
and NOTIFY messages. Make sure that the Connected Party Transformation CSS that
you choose contains the connected party transformation pattern that you want to
assign to this device.

Note

If you
configure the Connected Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the
Connected Party Transformation CSS in a non-null partition that is not used for
routing.

Use
Device Pool Connected Party Transformation CSS

To use
the Connected Party Transformation CSS that is configured in the device pool
that is assigned to this device, check this check box. If you do not check this
check box, the device uses the Connected Party Transformation CSS that you
configured for this device in the Trunk Configuration window.

Outbound
Calls

Called
Party Transformation CSS

This
setting allows you to send transformed called party number in SETUP message for
outgoing calls. Make sure that the Called Party Transformation CSS that you
choose contains the called party transformation pattern that you want to assign
to this device.

Note

If
you configure the Called Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the Called
Party Transformation pattern in a non-null partition that is not used for
routing.

Calling
Party Transformation CSS

This
setting allows you to send transformed calling party number in SETUP message
for outgoing calls. Also when redirection occurs for outbound calls, this CSS
will be used to transform the connected number sent from
Cisco Unified Communications Manager side in outgoing
NOTIFY messages. Make sure that the Calling Party Transformation CSS that you
choose contains the calling party transformation pattern that you want to
assign to this device.

Tip

If
you configure the Calling Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the Calling
Party Transformation Pattern in a non-null partition that is not used for
routing.

The
following table lists configuration settings for H.323 gateways.

Table 25 H.323
Gateway Configuration Settings

Field

Description

Connected Party Settings

Connected Party Transformation CSS

This
setting is applicable only for inbound Calls. This setting allows you to
transform the connected party number that
Cisco Unified Communications Manager sends in another
format, such as a DID or E.164 number. This setting is applicable while sending
connected number for basic call as well as sending connected number after
inbound call is redirected.

Cisco Unified Communications Manager includes the
transformed number in the Connected Number Information Element (IE) of CONNECT
and NOTIFY messages. Make sure that the Connected Party Transformation CSS that
you choose contains the connected party transformation pattern that you want to
assign to this device.

Note

If you
configure the Connected Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the
Connected Party Transformation CSS in a non-null partition that is not used for
routing.

Use
Device Pool Connected Party Transformation CSS

To use
the Connected Party Transformation CSS that is configured in the device pool
that is assigned to this device, check this check box. If you do not check this
check box, the device uses the Connected Party Transformation CSS that you
configured for this device in the Trunk Configuration window.

Call
Routing Information - Outbound Calls

Called
Party Transformation CSS

This
setting allows you to send transformed called party number in SETUP message for
outgoing calls. Make sure that the Called Party Transformation CSS that you
choose contains the called party transformation pattern that you want to assign
to this device.

Note

If you
configure the Called Party Transformation CSS as None, the transformation does
not match and does not get applied. Ensure that you configure the Called Party
Transformation CSS in a non-null partition that is not used for routing.

Calling
Party Transformation CSS

This
setting allows you to send transformed calling party number in SETUP message
for outgoing calls. Also when redirection occurs for outbound calls, this CSS
will be used to transform the connected number sent from
Cisco Unified Communications Manager side in outgoing
NOTIFY messages. Make sure that the Calling Party Transformation CSS that you
choose contains the calling party transformation pattern that you want to
assign to this device.

Note

If you
configure the Calling Party Transformation CSS as None, the transformation does
not match and does not get applied. Ensure that you configure the Calling Party
Transformation Pattern in a non-null partition that is not used for routing.

This
setting allows you to send transformed called party number in SETUP message for
outgoing calls. Make sure that the Called Party Transformation CSS that you
choose contains the called party transformation pattern that you want to assign
to this device.

Note

If
you configure the Called Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the Called
Party Transformation pattern in a non-null partition that is not used for
routing.

Calling
Party Transformation CSS

This
setting allows you to send transformed calling party number in SETUP message
for outgoing calls. Also when redirection occurs for outbound calls, this CSS
will be used to transform the connected number sent from
Cisco Unified Communications Manager side in outgoing
NOTIFY messages. [ For PRI DMS - 100 and DMS - 200 ]. Make sure that the
Calling Party Transformation CSS that you choose contains the calling party
transformation pattern that you want to assign to this device.

Tip

If
you configure the Calling Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the Calling
Party Transformation Pattern in a non-null partition that is not used for
routing.

Connected Party Settings

Connected Party Transformation CSS

This
setting is applicable only for inbound Calls. This setting allows you to
transform the connected party number sent from
Cisco Unified Communications Manager in another
format, such as a DID or E.164 number.

Note

You
can configure a Connected Party Transformation CSS only when you select one of
the following protocols that support Connected Number Information Element:

For T1 PRI :

PRI DMS - 100

PRI DMS - 250

PRI ISO QSIG T1

For E1 PRI :

PRI ISO QSIG E1

For
other protocol types, Connected Party Transformation CSS is grayed out.

Using
this setting,
Cisco Unified Communications Manager includes
transformed number in Connected Number Information Element ( IE) of CONNECT
message for basic call. For PRI DMS - 100 and DMS - 250 protocols ,
Cisco Unified Communications Manager includes
transformed number in Connected Number Information Element ( IE) of NOTIFY
message for inbound calls after redirection. Make sure that the Connected Party
Transformation CSS that you choose contains the connected party transformation
pattern that you want to assign to this device.

Note

If you
configure the Connected Party Transformation CSS as None, the transformation
does not match and does not get applied. Ensure that you configure the
Connected Party Transformation CSS in a non-null partition that is not used for
routing.

Use
Device Pool Connected Party Transformation CSS

To use
the Connected Party Transformation CSS that is configured in the device pool
that is assigned to this device, check this check box. If you do not check this
check box, the device uses the Connected Party Transformation CSS that you
configured for this device in the Trunk Configuration window.

Incoming
Called Party Settings

Clear
Prefix Settings

To
delete all prefixes for all called party number types, click Clear Prefix
Settings.

Default
Prefix Settings

To enter
the default value for all prefix fields at the same time, click Default Prefix
Settings.

National
Number

Configure the following settings to transform incoming called
party numbers that use National for the Called Party Number Type.

Prefix—Cisco
Unified Communications Manager applies the prefix that you enter in
this field to called party numbers that use National for the Called Party
Number Type. You can enter up to 16 characters, which include digits, the
international escape character (+), asterisk (*), or the pound sign (#). You
can enter the word, Default, instead of entering a prefix.

Tip

If
the word, Default, displays in the Prefix field, you cannot configure the Strip
Digits field. In this case,
Cisco Unified
Communications Manager takes the configuration for the Prefix and
Strip Digits fields from the device pool that is applied to the device. If the
word, Default, displays in the Prefix field in the Device Pool Configuration
window,
Cisco Unified
Communications Manager does not apply any prefix or strip digit
functionality.

To
configure the Strip Digits field, you must leave the Prefix field blank or
enter a valid configuration in the Prefix field. To configure the Strip Digits
fields, do not enter the word, Default, in the Prefix field.

Strip Digits—Enter the
number of digits that you want
Cisco Unified
Communications Manager to strip from the called party number of
National type before it applies the prefixes.

Use Device Pool CSS—Check
this check box to use the calling search space for the National Number field
that is configured in the device pool that is applied to the device.

Calling Search Space—This
setting allows you to transform the called party number of National called
party number type on the device. If you choose None, no transformation occurs
for the incoming called party number. Make sure that the calling search space
that you choose contains the called party transformation pattern that you want
to assign to this device.

International Number

Configure the following settings to transform incoming called
party numbers that use International for the Called Party Number Type.

Prefix—Cisco
Unified Communications Manager applies the prefix that you enter in
this field to called party numbers that use International for the Called Party
Numbering Type. You can enter up to 16 characters, which include digits, the
international escape character (+), asterisk (*), or the pound sign (#). You
can enter the word, Default, instead of entering a prefix.

Tip

If
the word, Default, displays in the Prefix field, you cannot configure the Strip
Digits field. In this case,
Cisco Unified
Communications Manager takes the configuration for the Prefix and
Strip Digits fields from the device pool that is applied to the device. If the
word, Default, displays in the Prefix field in the Device Pool Configuration
window,
Cisco Unified
Communications Manager does not apply any prefix or strip digit
functionality.

Tip

To
configure the Strip Digits field, you must leave the Prefix field blank or
enter a valid configuration in the Prefix field. To configure the Strip Digits
fields, do not enter the word, Default, in the Prefix field.

Strip Digits—Enter the
number of digits that you want
Cisco Unified
Communications Manager to strip from the called party number of
International type before it applies the prefixes.

Use Device Pool CSS—
Check this check box to use the calling search space for the International
Number field that is configured in the device pool that is applied to the
device.

Calling Search Space—This
setting allows you to transform the called party number of International called
party number type on the device. If you choose None, no transformation occurs
for the incoming called party number. Make sure that the calling search space
that you choose contains the called party transformation pattern that you want
to assign to this device.

Unknown
Number

Configure the following settings to transform incoming called
party numbers that use Unknown for the Called Party Number Type.

Prefix—Cisco
Unified Communications Manager applies the prefix that you enter in
this field to called numbers that use Unknown for the Called Party Numbering
Type. You can enter up to 16 characters, which include digits, the
international escape character (+), asterisk (*), or the pound sign (#). You
can enter the word, Default, instead of entering a prefix.

Tip

If
the word, Default, displays in the Prefix field, you cannot configure the Strip
Digits field. In this case,
Cisco Unified
Communications Manager takes the configuration for the Prefix and
Strip Digits fields from the device pool that is applied to the device. If the
word, Default, displays in the Prefix field in the Device Pool Configuration
window,
Cisco Unified
Communications Manager does not apply any prefix or strip digit
functionality.

Tip

To
configure the Strip Digits field, you must leave the Prefix field blank or
enter a valid configuration in the Prefix field. To configure the Strip Digits
fields in these windows, do not enter the word, Default, in the Prefix field.

Strip Digits—Enter the
number of digits that you want
Cisco Unified
Communications Manager to strip from the called party number of
Unknown type before it applies the prefixes.

Use Device Pool CSS—Check
this check box to use the calling search space for the Unknown Number field
that is configured in the device pool that is applied to the device.

Calling Search Space—This
setting allows you to transform the called party number of Unknown called party
number type on the device. If you choose None, no transformation occurs for the
incoming called party number. Make sure that the calling search space that you
choose contains the called party transformation pattern that you want to assign
to this device.

Subscriber Number

Configure the following settings to transform incoming called
party numbers that use Subscriber for the Called Party Number Type.

Prefix—Cisco
Unified Communications Manager applies the prefix that you enter in
this field to called numbers that use Subscriber for the Called Party Numbering
Type. You can enter up to 16 characters, which include digits, the
international escape character (+), asterisk (*), or the pound sign (#).You can
enter the word, Default, instead of entering a prefix.

Tip

If
the word, Default, displays in the Prefix field, you cannot configure the Strip
Digits field. In this case,
Cisco Unified
Communications Manager takes the configuration for the Prefix and
Strip Digits fields from the device pool that is applied to the device. If the
word, Default, displays in the Prefix field in the Device Pool Configuration
window,
Cisco Unified
Communications Manager does not apply any prefix or strip digit
functionality.

Tip

To
configure the Strip Digits field, you must leave the Prefix field blank or
enter a valid configuration in the Prefix field. To configure the Strip Digits
fields, do not enter the word, Default, in the Prefix field.

Strip Digits—Enter the
number of digits that you want
Cisco Unified
Communications Manager to strip from the called party number of
Subscriber type before it applies the prefixes.

Use Device Pool CSS—Check
this check box to use the calling search space for the Subscriber Number field
that is configured in the device pool that is applied to the device.

Calling Search Space—This
setting allows you to transform the called party number of Subscriber called
party number type on the device. If you choose None, no transformation occurs
for the incoming called party number. Make sure that the calling search space
that you choose contains the called party transformation pattern that you want
to assign to this device.

To
delete all prefixes for all called party number types, click Clear Prefix
Settings.

Default
Prefix Settings

To enter
the default value for all prefix fields at the same time, click Default Prefix
Settings.

National
Number

Configure the following settings to transform incoming called
party numbers that use National for the Called Party Number Type.

Prefix—Cisco Unified Communications Manager applies the
prefix that you enter in this field to called party numbers that use National
for the Called Party Numbering Type. You can enter up to 16 characters, which
include digits, the international escape character (+), asterisk (*), or the
pound sign (#). You can enter the word, Default, instead of entering a prefix.

Tip

If the word, Default, displays in the Prefix field, you cannot
configure the Strip Digits field. In this case,
Cisco Unified
Communications Manager takes the configuration for the Prefix and
Strip Digits fields from the device pool that is applied to the device. If the
word, Default, displays in the Prefix field in the Device Pool Configuration
window,
Cisco Unified
Communications Manager applies the service parameter configuration
for the incoming called party prefix, which supports both the prefix and strip
digit functionality.

Tip

To configure the Strip Digits field, you must leave the Prefix
field blank or enter a valid configuration in the Prefix field. To configure
the Strip Digits fields, do not enter the word, Default, in the Prefix field.

Strip Digits—Enter the
number of digits that you want Cisco Unified Communications Manager to strip
from the called party number of National type before it applies the prefixes.

Use Device Pool CSS—
Check this check box to use the calling search space for the National Number
field that is configured in the device pool that is applied to the device.

Calling Search Space—This
setting allows you to transform the called party number of National called
party number type on the device. If you choose None, no transformation occurs
for the incoming called party number. Make sure that the calling search space
that you choose contains the called party transformation pattern that you want
to assign to this device.

International Number

Configure the following settings to transform incoming called
party numbers that use International for the Called Party Number Type.

Prefix—Cisco Unified Communications Manager applies the
prefix that you enter in this field to called party numbers that use
International for the Called Party Numbering Type. You can enter up to 16
characters, which include digits, the international escape character (+),
asterisk (*), or the pound sign (#). You can enter the word, Default, instead
of entering a prefix.

Tip

If the word, Default, displays in the Prefix field, you cannot
configure the Strip Digits field. In this case,
Cisco Unified
Communications Manager takes the configuration for the Prefix and
Strip Digits fields from the device pool that is applied to the device. If the
word, Default, displays in the Prefix field in the Device Pool Configuration
window,
Cisco Unified
Communications Manager applies the service parameter configuration
for the incoming called party prefix, which supports both the prefix and strip
digit functionality.

Tip

To configure the Strip Digits field, you must leave the Prefix
field blank or enter a valid configuration in the Prefix field. To configure
the Strip Digits fields, do not enter the word, Default, in the Prefix field.

Strip Digits—Enter the
number of digits that you want Cisco Unified Communications Manager to strip
from the called party number of International type before it applies the
prefixes.

Use Device Pool CSS—Check
this check box to use the calling search space for the International Number
field that is configured in the device pool that is applied to the device.

Calling Search Space—This
setting allows you to transform the called party number of International called
party number type on the device. If you choose None, no transformation occurs
for the incoming called party number. Make sure that the calling search space
that you choose contains the called party transformation pattern that you want
to assign to this device.

Unknown
Number

Configure the following settings to transform incoming called
party numbers that use Unknown for the Called Party Number Type.

Prefix—Cisco Unified Communications Manager applies the
prefix that you enter in this field to called numbers that use Unknown for the
Called Party Numbering Type. You can enter up to 16 characters, which include
digits, the international escape character (+), asterisk (*), or the pound sign
(#). You can enter the word, Default, instead of entering a prefix.

Tip

If the word, Default, displays in the Prefix field, you cannot
configure the Strip Digits field. In this case,
Cisco Unified
Communications Manager takes the configuration for the Prefix and
Strip Digits fields from the device pool that is applied to the device. If the
word, Default, displays in the Prefix field in the Device Pool Configuration
window,
Cisco Unified
Communications Manager applies the service parameter configuration
for the incoming called party prefix, which supports both the prefix and strip
digit functionality.

Tip

To configure the Strip Digits field, you must leave the Prefix
field blank or enter a valid configuration in the Prefix field. To configure
the Strip Digits fields in these windows, do not enter the word, Default, in
the Prefix field.

Strip Digits—Enter the
number of digits that you want Cisco Unified Communications Manager to strip
from the called party number of Unknown type before it applies the prefixes.

Use Device Pool CSS—Check
this check box to use the calling search space for the Unknown Number field
that is configured in the device pool that is applied to the device.

Calling Search Space—This
setting allows you to transform the called party number of Unknown called party
number type on the device. If you choose None, no transformation occurs for the
incoming called party number. Make sure that the calling search space that you
choose contains the called party transformation pattern that you want to assign
to this device.

Subscriber Number

Configure the following settings to transform incoming called
party numbers that use Subscriber for the Called Party Number Type.

Prefix—Cisco Unified Communications Manager applies the
prefix that you enter in this field to called numbers that use Subscriber for
the Called Party Numbering Type. You can enter up to 16 characters, which
include digits, the international escape character (+), asterisk (*), or the
pound sign (#). You can enter the word, Default, instead of entering a prefix.

Tip

If the word, Default, displays in the Prefix field, you cannot
configure the Strip Digits field. In this case,
Cisco Unified
Communications Manager takes the configuration for the Prefix and
Strip Digits fields from the device pool that is applied to the device. If the
word, Default, displays in the Prefix field in the Device Pool Configuration
window,
Cisco Unified
Communications Manager applies the service parameter configuration
for the incoming called party prefix, which supports both the prefix and strip
digit functionality.

Tip

To configure the Strip Digits field, you must leave the Prefix
field blank or enter a valid configuration in the Prefix field. To configure
the Strip Digits fields, do not enter the word, Default, in the Prefix field.

Strip Digits—Enter the
number of digits that you want Cisco Unified Communications Manager to strip
from the called party number of Subscriber type before it applies the prefixes.

Use Device Pool CSS—Check
this check box to use the calling search space for the Subscriber Number field
that is configured in the device pool that is applied to the device.

Calling Search Space—This
setting allows you to transform the called party number of Subscriber called
party number type on the device. If you choose None, no transformation occurs
for the incoming called party number. Make sure that the calling search space
that you choose contains the called party transformation pattern that you want
to assign to this device.

Add Route Groups to Route List

Note

When you
configure the Local Route Group feature, add the route groups to the route list
by selecting those local route group names that are appended with the Local
Route Group tag that appears in the drop-down list box.

IP Phones
Considerations

RTMT
Considerations

Security
Considerations

Serviceability
Considerations

About Local Route Group Names Setup

In Cisco Unified Communications Manager Administration, use the Call Routing > Route/Hunt > Local Route Group Names menu path to configure local route group names.

A local route group name is a unique name that you assign to a local route group in the Local Route Group Names window. The Local Route Group Names window allows you to add and configure multiple local route group names that you can customize and associate with route groups for a given device pool.

Local Route Group Names Settings

The following table describes the available fields and buttons in the Local Route Group Names window.

Table 28 Local Route Group Names Settings

Field or Button

Description

Name

Enter a unique local route group name in this required field. The name can comprise up to 50 alphanumeric characters and can contain any combination of spaces, periods (.), hyphens (-), and underscores (_).

Note

The Standard Local Route Group entry in the Name field is a default entry. It is populated from pre-10.0(1) release input. This field is editable. It allows you to change the name to a name of your choice.

Note

The Device Pool Configuration window under System > Device Pool displays the local route group name entries as labels under Local Route Group Settings.

Description

(Optional) Enter a description that will help you to distinguish between local route group names. You can change the description if required. The description can comprise up to 100 alphanumeric characters except the following characters: ampersand (&), double quotation marks ("), angle brackets (<>), and percent (%).

Add Row

Click this button to add new local route group names. This button adds an empty row below the previous row entry. Enter the name and description of the local route group that you want to add to this row.

To delete an existing local route group name, click the Minus Sign (-) button at the right corner of the relevant row. Click Save to confirm the process.

Note

By default, the Minus Sign (-) button in the first row is inactive.

Note

You can delete an existing local route group name only if it does not have any dependency on any device pool
or route list. To delete an existing local route group, you must first find the associated device pools as well as the route lists from the dependency record, disassociate them, and then delete the local route group name.

Save

Click this button to save the local route group name entries.

SAML Single Sign On

Cisco Unified Communications Manager Administration Considerations

The Security Assertion Markup Language (SAML) Single Sign On feature allows end users to log into a Windows client machine and then access certain Cisco Unified Communications Manager applications without logging in again.

After you enable SAML Single Sign On (SSO), users are able to access the following web applications without logging in again:

Cisco Unified Communications Manager Administration

Cisco Unified Reporting

Cisco Unified Serviceability

If the Windows desktop authentication for SSO is configured for an end user, the end user is able to access all the above web applications without logging in again. However, if the Windows desktop SSO authentication is not configured, when the end users attempt to log into a SAML-enabled web
application, they are redirected to their configured
Identity Provider (IdP) to enter the authentication details. After
successful authentication by the IdP, the web browser redirects the users to the web application that they were trying to access.

At least one
LDAP synchronized user is added to the Standard CCM Super Users group to enable
access to Cisco Unified Administration.

Note

For more
information about synchronizing end-user data and adding LDAP-synchronized
users to a group, see the "System setup" and "End user setup" sections in the
Cisco
Unified Communications Manager Administration Guide.

OpenAM SSO
(Cisco Unified OS
Administration > Security > Single Sign
On or
Cisco Unified IM and
Presence OS Administration > Security > Single Sign
On) is disabled on all the nodes. For information
about OpenAM SSO, see
Single Sign-On and the
Deployment Guide for IM and Presence Service on Cisco Unified
Communications Manager.

A warning
message is displayed to notify you that all server connections will be
restarted.

Step 3

Click
Continue.

A dialog box
that allows you to import IdP metadata displays. To configure the trust
relationship between the IdP and your servers, you must obtain the trust
metadata file from your IdP and import it to all your servers.

Step 4

Click
Browse to
locate and upload the IdP metadata file.

Step 5

Click
Import IdP
Metadata.

Step 6

Click
Next.

Note

The
Next button is enabled only if the IdP metadata file
is successfully imported on at least one node in the cluster.

After you
install the server metadata on the IdP server, you must run an SSO test to
ensure that the metadata files are correctly configured.

Step 9

Click
Next to
continue.

Step 10

Select an
LDAP-synced user with administrator rights from the list of valid administrator
IDs.

Step 11

Click
Run Test.

The IdP login
window displays.

Note

You cannot
enable SAML SSO until the Run Test succeeds.

Step 12

Enter a valid
username and password.

After
successful authentication, the following message is displayed:

SSO Test Succeeded

Close the
browser window after you see this message.

If the
authentication fails or takes more than 60 seconds to authenticate, a "Login
Failed" message is displayed on the IdP login window. The following message is
displayed on the SAML Single Sign-On window:

SSO Metadata Test Timed Out

To attempt
logging in to the IdP again, repeat Steps 11 and 12.

Step 13

Click
Finish to
complete the SAML SSO setup.

SAML SSO is
enabled and all the web applications participating in SAML SSO are restarted.
It may take one to two minutes for the web applications to restart.

Security-By-Default Enhancements

Security-by-Default (SBD) is a feature that allows Cisco Unified
Communications Manager users to benefit from security features out of the box
without the user having to perform configuration tasks.

SBD leverages the
ITL file, which is a digitally signed file that contains all Unified
Communications Manager node certificates (including TVS) that endpoints can
trust.

The SBD
enhancements address the issues and gaps that can lead to phones getting locked
and customers not having to delete the ITL file either manually from the phones
or by using third party developed tools.

In Unified
Communications Manager 10.0(1), a new ITL reset private key called ITLRecovery
is regenerated and is accessible from the Certificate Management interface.

The ITL reset
certificate is pushed into the database with SAST role, which allows TFTP nodes
to find this certificate when they build the ITL file. TFTP nodes then include
this record in the ITL file.

Cisco Unified
Communications Manager Administration Considerations

No changes.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

When phones are in
a locked state and they are not accepting any ITL or configuration changes from
their Unified Communications manager cluster, a new CLI command
utils ITL
reset can be used to create a special ITL Recovery ITL file. This
process takes the existing ITL file from the publisher node, strips the
signature of the ITL file and signs the contents of the ITL file again with the
unlocked reset ITL Recovery private key.

The new ITL file
is served to the TFTP directories on all the active TFTP nodes in the cluster.
TFTP services are restarted. After this command is run successfully, the
administrator must manually restart all the phones from the GUI. See
"Perform bulk
reset of ITL file."

Note

For the bulk ITL
file reset to work, the ITL Recovery certificate and key need to be available.
You must back up this file.

Serviceability
Considerations

Perform Bulk Reset of ITL File

When devices on a Unified Communications Manager cluster are locked and lose their trusted status, perform a bulk reset of the Identity Trust List (ITL) file with the CLI command utils itl reset. This command generates a new ITL recovery file.

Tip

Whenever you perform a fresh installation of Unified Communications Manager, export the ITL key as soon as possible and perform a backup through the Disaster Recovery System.

The CLI command to export the ITL recovery pair
is as follows:

file get tftpITLRecovery.p12

You will be prompted to enter the SFTP server (where the key will be exported) and
password.

Before You Begin

Make sure you perform this procedure on the Cisco Unified Communications Manager publisher.

If needed, export the key from the publisher.

Procedure

Step 1

Perform one of the following steps:

Run utils itl reset localkey.

Run utils itl reset remotekey.

For utils itl reset localkey, the local key resides on the publisher. This step generates a new ITL file by taking the existing file on the system and replacing the signature of that file with the recovery key signature. The key is then copied to the TFTP servers in the cluster.

The devices restart. They are ready to download the ITL file that is signed by the ITLRecovery key and accept configuration files.

Step 5

Restart the TFTP service and restart all devices.

The devices download the ITL file that is signed with the TFTP key and register correctly to Unified Communications Manager again.

CLI changes

utils itl reset

This command is used when endpoints are unable to validate their configuration files.

utils itl reset
{ localkey | remotekey }

Syntax Description

localkey

Generates a new ITL file by taking the existing ITL file on the publisher. The command replaces the signature of that ITL file and signs the new ITL file with the ITL recovery key.

remotekey

Generates a new ITL file after importing the PKCS 12 bag that contains the recovery certificate key pair from the remote location. It then signs the newly generated ITL file with the recovery private key.

remotekey has the following parameters:

IP address or hostname

User ID

ITLRecovery.p12

Command Modes

Administrator (admin:)

Usage Guidelines

Note

You must run this command on the Unified Communications Manager publisher node.

Requirements

Command privilege level: 4

Allowed during upgrade: No

Applies to: Unified Communications Manager

Example

admin:utils itl reset
Name is None
Generating the reset ITL file.....
The reset ITL file was generated successfully
Locating active Tftp servers in the cluster.....
Following is the list of Active tftp servers in the cluster
====================
se032c-94-42
=====================
Number of Active TFTP servers in the cluster : 1
Transferring new reset ITL file to the TFTP server nodes in the cluster.........
Successfully transferred reset ITL to node se032c-94-42

Additional information

Proxy TFTP and Security

Endpoints in a Cisco Unified Communications Manager cluster are
configured with Proxy TFTP (for example, through Dynamic Host Configuration Protocol, or DHCP). Proxy TFTP can find the target cluster of the
endpoint.

Note

Cisco recommends that you keep the Proxy TFTP
on the current release while you upgrade the rest of the clusters, as well as have a combination of nonsecure and mixed-mode
clusters.

The Proxy TFTP server does not have to be on the
highest Unified Communications Manager release, and clusters in a Proxy TFTP deployment can be
either nonsecure or in mixed-mode.

Proxy TFTP
can find the target cluster of endpoints because the MAC address
of the endpoints is part of the filename in the TFTP GET request (for example,
SEP001956A3A472.cnf.xml.sgn). Proxy TFTP discovers the target in the following way:

Proxy TFTP polls all the
clusters that it controls for the requested file, starting from its own
database.

The cluster where the endpoint is configured returns
the file.

The locale and ring list file
requests do not contain a MAC address, so Proxy TFTP returns its own copies of these files.

When Security-by-Default (SBD) was introduced for Unified Communications Manager, Proxy TFTP (and
TFTP servers in general) served both signed and nonsigned
requests.

If the home cluster of an
endpoint does not accept the ITL file request, the
endpoint requests a default ITL file which the Proxy TFTP serves. After the endpoint receives the configuration file from its home cluster, the endpoint cannot validate the signature, because the endpoint has the ITL file from the Proxy TFTP and not its home cluster.

To address this issue, the TFTP service returns a file not found message when the
default ITL file is requested.

10.0(1) Proxy TFTPs perform the following steps for signing files and serving them to endpoints:

Automatically discover the cluster in the
deployment that is on the highest release

Get the locale and ring
list files from the cluster

Strip the signature of the locale or ring list file

Sign the files with
their own TFTP private key before serving them to endpoints that are requesting the files

Self Care User Options

Unified Communications Self Care Portal is a new web-based user interface that phone users can access to set up user settings for their Cisco Unified IP Phones. Unified Communications Self Care Portal replaces the Cisco Unified CM User Options interface from the last release.

Cisco Unified Communications Manager Administration Considerations

As with Cisco Unified CM User Options from the last release, administrators can set enterprise parameters in Cisco Unified Communications Manager that control which settings are available for users to configure in the Self Care interface. These enterprise parameters appear in the Enterprise Parameter Configuration window under the User Options Parameters heading.

Bulk Administration Considerations

No changes.

CDR/CAR Considerations

No changes.

IP Phones Considerations

No changes.

RTMT Considerations

No changes.

Security Considerations

No changes.

Serviceability Considerations

No changes.

Self-Provisioning

Self-Provisioning for End Users and Administrators

The
Self-Provisioning feature allows an end user or administrator to add an
unprovisioned phone to a Cisco Unified Communications Manager system with
minimal administrative effort. A phone can be added by plugging it into the
network and following a few prompts to identify the user.

This feature
enhances the out-of-box experience for end users by allowing them to directly
add their desk phone or soft client without contacting the administrator. It
simplifies administrator deployments by allowing them to add desk phones on
behalf of an end user. The feature lets administrators and users deploy a large
number of devices without interacting directly with the Cisco Unified
Communications Manager Administration GUI, but from the device itself. The
feature relies on the administrator preconfiguring a number of templates and
profiles, so that when the phone attempts to self-provision, the necessary
information is available in the system for it to create a new device.

Note

Self-provisioning is not supported for secured endpoints.

There are two
levels of configuration for Self-Provisioning:

The system
level

The user level

You can set up
this feature at the system level from Cisco Unified Communications Manager
Administration under the
User
Management > Self-Provisioning menu.

To set up this
feature, you can select one of the following modes:

Secure
Mode

Administrators can provision devices on behalf of end users

End
users can provision devices with their credentials

Non-Secure Mode

End
users/administrators can enter Self-Service ID for the device that is being
provisioned.

With
appropriately configured User Profiles, end users can provision their own
devices. These User Profiles may be shared by a group of users that share the
same characteristics. The User Profile contains the following settings:

Universal
Device Templates

Universal Line
Template

End user
Self-Provisioning settings

Note

The
administrator can set any User Profile as the system default.

In order to allow
a user to provision a new device using Self-Provisioning, the user must meet
the following criteria:

If you do not
configure a UDT in the User Profile, user assignment fails and plays the
following error message on the phone:
This
device could not be associated to your account. Please contact the System
administrator to complete provisioning.

Self-Provisioning must be enabled for the end user.

Note

Self-Provisioning must be enabled even if the administrator
performs device self-provisioning on behalf of the user.

The user must
have a primary extension.

The user must
have the appropriate universal device template linked to the User Profile.

The total
number of owned devices must be less than the Self-Provisioning limit that is
specified on the associated User Profile.

Self-Provisioning IVR Service

The
Self-Provisioning feature introduces a new service called Self-Provisioning IVR
service. When you dial the CTI RP DN that is configured on the
Self-Provisioning page, from an extension of a user that uses the IVR service,
the phone connects to the Self-Provisioning IVR application and prompts you to
provide the Self-Service credentials. Based on the validation of the
Self-Service credentials that you provide, the IVR service assigns the
autoregistered IP phones to the users.

You can configure
self-provisioning even if the service is deactivated, but the administrator
cannot assign IP phones to users using the IVR service. By default, this
service is deactivated.

Note

When you upgrade
a previous release
Cisco
Unified Communications Manager to Release 10.0, the
Cisco
Unified Communications Manager will create a Universal Device
Template and a Universal Line Template which will retain the previous
configurations for Auto-Registration settings. After the upgrade, the values of
Partition and
External Phone Number Mask will be populated in the
new Universal Line Template by
Cisco
Unified Communications Manager and in the Line field of the Universal
Device Template respectively. And also, the
Cisco
Unified Communications Manager populates the
Cisco
Unified Communications Manager name for the Universal Device Template
and a Universal Line Template and configures the same values for
Auto-Registration settings.

Cisco Unified
Communications Manager Administration Considerations

For this feature,
the following are new GUI menu paths in Cisco Unified Communications Manager
Administration that allow you to configure Self Provisioning:

User
Management > Self-Provisioning

Allows you
to set up Self-Provisioning for endpoints and the Self-Provisioning IVR
service.

User
Management > User Settings > User
Profile

Allows you
to create User Profiles.

Note

The
Universal Device Template and Universal Line Template setup fields are now
available on this window.

The following GUI
menu items have been updated for the Self-Provisioning feature:

User
Management > User/Phone Add > Feature Group
Template

A User
Profile field has been added.

User
Management > User/Phone Add > Quick User/Phone
Add

The Feature
Group Template field has been moved to the User Information section.

For administrator authentication, specify the authentication code. The authentication code must be an integer ranging from 0 to 20 digits but cannot be empty (null).

Step 3

Find an existing user in the Unified Communications Manager database.

Step 4

Find the User Profile that is associated with the user.

Step 5

Open the User Profile.

Step 6

Check the Allow end user to provision their own phones check box.

Step 7

Select Save.

The user is now able to perform self-provisioning on the device.

Session
Description Protocol Transparency

The Session
Description Protocol (SDP) Transparency Profile can be configured to
selectively allow declarative parameters or to allow all unrecognized
parameters to pass from the ingress call leg to the egress call leg.

Session
Description Protocol Transparency for Declarative Parameters

The Session
Description Protocol Transparency for Declarative Parameters allows the
administrator to specify declarative SDP attributes that are not natively
supported by Cisco Unified Communications Manager (Unified Communications
Manager) to be passed from the ingress call leg to the egress call
leg. If the
Unified
Communications Manager receives attributes that are not explicitly
identified by the administrator to send to the egress leg,
Unified
Communications Manager drops the attribute from the outgoing SDP
similar to previous versions of
Unified
Communications Manager. This feature allows the administrator to
identify attributes that are sent to the egress leg in multiple ways, such as
configuring all property attributes with a particular name, all value
attributes with a particular name, or all value attributes with a specific name
and specific value to be passed through. The administrator can also configure
all unrecognized attributes to be passed along in the outgoing SDP.

Note

SDP
Transparency for Declarative Parameters only applies to declarative attributes,
not to negotiated attributes.

The
Cisco
Unified Communications Manager first looks at the name field of an
incoming attribute. If the default "Pass all unknown SDP attributes" profile is
not used,
Unified
Communications Manager looks for an exact match among the attributes
designated to be passed through. An exact match between the name field of the
attribute arriving on the ingress call leg and the name defined by the
administrator occurs only if the two strings are identical (case sensitive). If
an exact match is not found, then the attribute is not passed through.

The following are
the three attributes that can be configured:

Property
attributes: When an administrator configures a property attribute in the SDP
Transparency Profile, the attribute is passed through unless the incoming
attribute has a value. If the incoming attribute has a value,
Unified Communications Manager categorizes the
incoming attribute as a value attribute and it is not passed through.

Value
attributes: When an administrator configures a value attribute of any value in
the SDP Transparency Profile to be passed through, the attribute is passed
through if it contains a value that includes at least one non-white space
character (horizontal tab or space). If the value payload consists of all white
space characters,
Unified Communications Manager categorizes it as a
value attribute and it is not passed through.

Value
attributes configured for value from list: The attribute is passed through only
if the value matches one of the five specified values identified by the
administrator. If the value does not match one of the five specified values or
the there is no value, then the attribute is not passed through.

An administrator
can configure the SDP Transparency Profile to pass all unrecognized SDP
attributes from the ingress call leg to the egress call leg when the SDP
Transparency Profile is set to "Pass all unknown SDP attributes". To prevent
all unrecognized SDP attributes from passing through set the SDP Transparency
Profile to "None". The SDP Transparency Profile is selected as "Pass all
unrecognized SDP attributes" by default for:

Standard SIP
Profile for Cisco VCS

Standard SIP
Profile for Telepresence Conferencing

Standard SIP
Profile for Telepresence Endpoints

Limitations

Because of the
nature of the existing SDP parsing infrastructure that is shared by multiple
products, there are certain limitations in the ability to pass through an
attribute without any changes.

This feature does
not support attribute lines longer than XXXX chars inclusive of a, =, and CRLF.
To avoid the limitations, it is recommended that devices passing SDP to the
Unified
Communications Manager in the ingress leg conform to RFC 4566 which
define attribute syntax as:

a=<name> for property attributes

a=<name>:<value> for value attributes

Also avoid errors
that can result from using non-standard attribute formatting. Even though
adherence to RFC 4566 is not required to use this feature, devices following
RFC 4566 are immune from the limitations discussed above.

The Find and List SDP Transparency Profile page lists all available SDP Transparency Profiles. You may need to click Find or Clear Filter if no SDP Transparency Profiles appear on the list. This list may also contain several SDP Transparency pre-configured profiles that come with Unified Communications Manager. These profiles may be copied and modified to suit your needs.

Step 2

Perform one of the following:

Select Add New to create a new SDP Transparency Profile.

Open an existing SDP Transparency Profile.

Note

You cannot edit the Pass all unknown SDP attributes profile.

Step 3

Enter the Profile Information.

See the SDP Transparency Profile settings table.

Step 4

Enter the Attribute Information.

See the SDP Transparency Profile settings table.

Step 5

Click Save.

After the SDP Transparency Profile is ready, it needs to be associated with a SIP Profile.

On the SIP Profile page, select the desired SDP Transparency Profile from the drop-down list box

Step 8

Click Save.

Devices using the SIP Profile must be reset for the changes to take effect.

Note

The administrator can configure that all unrecognized attributes are passed to the egress leg by selecting the preconfigured SDP Transparency Profile named Pass all unknown SDP attributes from the SDP Transparency Profile drop-down list box. No other configuration is needed to pass through any unrecognized attribute.

Session Description Protocol Transparency Profile Settings

The following table describes the available fields in the
SDP Profile window.

Table 29 SDP Transparency Profile Settings

Field

Description

Profile Information

Name

The name of the SDP Transparency Profile

Note

The name must be unique among all SDP Transparency Profiles in the cluster.

Description

The administrator may also include an additional description about this particular profile

Attribute
Information

Name

Name of the attribute that is passed through

Type

Any Value: signifies that the attribute is passed through regardless of the value

Note

If the attribute is not a value attribute, it is not passed through.

Property: signifies that the attribute is a property attribute and therefore does not have a value

Example: a=foo In this example "foo" represents the property to be passed through.

Value From List: signifies that attributes that only contain specified values are passed through

Note

The administrator is limited to specifying up to five different values.

Example: a=foo:bar In this example foo represents the field, and bar represents one of the values that can be assigned.

Session Persistency

Roam between different networks (e.g. Wi-Fi, VPN over 3G/4G) without having to re-register with Cisco Unified Communications Manager (Unified Communications Manager).

Maintain the SIP-based subscription status with Unified Communications Manager while roaming between different networks.

Maintain registration with Unified Communications Manager in the case of network connectivity loss.

Seamlessly transit both active and held calls from one network to another without call drops.

To facilitate connectivity during roaming between networks, Session Persistency allows dynamic IP address/port change via keep-alive registration to facilitate connectivity during roaming between networks. In addition, the feature includes a configurable TCP reconnect timer, which must be enabled at the product level, to allow mobile users to remain connected in case of a temporary network connectivity loss or roaming. The timer is in effect only when the mobile device tears down the original TCP connection explicitly.

Cisco Unified Communications Manager Administration Considerations

If the TCP reconnect timer has been enabled at the product level, you can configure the timer by setting a value for the Time to Wait for Seamless Reconnect After TCP Drop or Roaming field. The field has a range of 0-300 seconds with a default value of five seconds. The field can be set from any of the following configuration windows:

Phone Configuration window

Common Phone Profile window

Enterprise Phone Configuration window

Bulk Administration Considerations

No changes.

CDR/CAR Considerations

No changes.

IP Phones Considerations

No changes.

RTMT Considerations

No changes.

Security Considerations

No changes.

Serviceability Considerations

No changes.

Session Timer With
Update

Session
Refresh Method

The session
refresh timer allows for periodic refresh of SIP sessions, which allows the
Unified Communications Manager and remote agents to determine whether the SIP
session is still active. Prior to Release 10.0(1), when the Unified
Communications Manager received a refresh command, it supported receiving
either Invite or Update SIP requests to refresh the session. When the Unified
Communications Manager initiated a refresh, it supported sending only Invite
SIP requests to refresh the session. With Release 10.01, this feature extends
the refresh capability so that Unified Communications Manager can send both
Update and Invite requests.

Cisco Unified
Communications Manager Administration Considerations

The following row
is added to the Session profile settings table:

Session
Refresh Method

Specify
whether Invite or Update should be used as the Session Refresh Method.

Invite (default)

Note

Sending a mid-call Invite request requires that an offer SDP be
specified in the request. This means that the far end must send an answer SDP
in the Invite response.

Update: Unified
Communications Manager sends a SIP Update request, if support for the Update
method is specified by the far end of the SIP session either in the Supported
or Require headers. When sending the Update request, the Unified Communications
Manager includes an SDP. This simplifies the session refresh since no SDP
offer/answer exchange is required.

Note

If the
Update method is not supported by the far end of the SIP session, the Unified
Communications Manager continues to use the Invite method for session refresh.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Single-Step IP Address and Hostname Change

Cisco Unified Communications Manager (Unified Communications Manager) has been updated with a simplified procedure for updating the IP address or hostname of a Unified Communications Manager server. For details, see the Changing the IP Address and Hostname in Cisco Unified Communications Manager, Release 10.0(1).

Cisco Unified Communications Manager Administration Considerations

The IP address and hostname of a Unified Communications Manager publisher or subscriber node can be updated from Cisco Unified Operating System Administration, or from the Command Line Interface.

Bulk Administration Considerations

No changes.

CDR/CAR Considerations

No changes.

IP Phones Considerations

No changes.

RTMT Considerations

No changes.

Security Considerations

No changes.

Serviceability Considerations

No changes.

Universal Line
Template

The Universal Line Template (ULT) feature allows you to create templates with settings that you would normally apply to a directory number. You can create one or more ULTs to reflect your most common directory number configurations, and apply the templates when adding a new directory number on the Quick User/Phone Add window.

Tip

To make the window easier to view, the template sections are
collapsed by default. Expand sections that you need as you walk
through the template setup process. Select the
Expand All
button to expand all sections.

Note

The ULT sections in this window may appear in
a different order than the settings table indicates. To change the
order of these settings, use the
User Management > User/Phone Add > Page Layout Preference menu.

Cisco Unified
Communications Manager Administration Considerations

You can find ULT
under the following GUI menu item:
User
Management > User/Phone Add > Universal Line
Template

The following GUI
menu items have been updated for the ULT feature:

User
Management > User/Phone Add > Page Layout
Preferences

A ULT
link has been added, where you can customize the layout of the ULT window.

User
Management > User/Phone Add > Feature Group
Template

A ULT
section has been added, which contains a drop-down list box from which you can
select a ULT. You can also click View Details to display settings for the ULT
that you selected.

User
Management > User/Phone Add > Quick User/Phone
Add

Under
the Extensions section when you click New, you see a Line Template drop-down
list box, from which you can select a ULT.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

User Agent and
Server Headers

User-Agent and
Server Header Information

This feature
provides the option to configure, in the SIP profile, the portion of the
installed build number that is sent in SIP messages. This value is used to
populate the User-Agent header in SIP requests and the Server header in SIP
responses.

Cisco Unified
Communications Manager Administration Considerations

The following row
is added to the SIP profile settings table:

User-Agent and Server header information

This
feature indicates how
Unified Communications
Manager handles the User-Agent and Server header information in a SIP
message.

Choose
one of the following three options:

Send Unified Communications
Manager Version Information as User-Agent Header—For INVITE requests, the
User-Agent header is included with the CM version header information. For
responses, the Server header is omitted.
Unified Communications
Manager passes through any contact headers untouched. This is the
default behavior.

Pass Through Received
Information as Contact Header Parameters —If this option is selected, the
User-Agent/Server header information is passed as Contact header parameters.
The User-Agent/Server header is derived from the received Contact header
parameters, if present. Otherwise, they are taken from the received
User-Agent/Server headers.

Pass Through Received
Information as User-Agent and Server Header—If this option is selected, the
User-Agent/Server header information is passed as User-Agent/Server headers.
The User-Agent/Server header is derived from the received Contact header
parameters, if present. Otherwise, they are taken from the received
User-Agent/Server headers.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Video Codec
Support

Unified
Communications Manager maintains the offerer' s video codec ordering preference
when making the answer, if possible. H.265 is the preferred video codec if
available on the endpoints, otherwise, Unified Communications Manager follows
the following codec preference order:

H.265 (HEVC):
provides higher quality video using lower bandwidth.

H.264 (SVC):
allows rendering of variable quality video from the same media stream, by
disregarding a subset of the packets received.

H.264 (AVC)
Advanced Video Coding

H.263

H.261

H. 264 SVC is a
new annex to H.264-AVC video compression standard; meaning it is an enhancement
on top of H.264-AVC. It provides the ability to encapsulate multiple video
streams at various frame-rates and resolutions in one container.

Video on
Hold

The Video on Hold
feature is for video contact centers where customers that place a call are able
to watch a specific video after initial consultation with the contact center
agent. In this case, the agent selects the video stream that is played to the
customer while the customer is on hold.

In addition to
the video contact center Video on Hold can be deployed within any enterprise if
the deployment requires a generic video on hold capability.

Cisco Unified
Communications Manager (Unified Communications Manager) now has a new
configuration "Video on Hold Server" that allows a media content server to be
provisioned under the existing "Media Resources" menu. The media content server
can stream audio and video content when directed by Unified Communications
Manager. The media content server is an external device that can store and
stream audio and video content under Unified Communications Manager control
using SIP as the signal protocol. The media content server is capable of
providing hi-definition video content at 1080p, 720p, or lower resolutions such
as 360p.

In addition to
the video contact centre, Video on Hold can be deployed within any enterprise
if the deployment requires a generic video on hold capability. The configured
Default Video Content Identifier for the Video on Hold server is used to play
the video stream to the user on hold.

The media content
server configuration and allocation for a particular session follows the "Media
Resource Group" and "Media Resource Group List" constructs in Unified
Communications Manager.

Cisco MediaSense
is used as the media content server.

Interaction
with Enhanced Location Call Admission Control

For this feature,
the Cisco MediaSense servers can be collocated in a Unified Communications
Manager cluster (the Cisco MediaSense cluster is directly connected to the
cluster where the holding party is registered). In that case, the Unified
Communications Manager cluster is responsible for deducting the bandwidth
between the location of the party on hold and the Cisco MediaSense location.
Since Video on Hold interactions make use of 720p or 1080p video streams, it is
important to take the bandwidth usage into account before allowing new sessions
in order to maintain video quality of existing sessions.

Video on Hold
Setup

Configure Unified
Communications Manager with a SIP trunk to a Cisco MediaSense cluster. The SIP
trunk to the Cisco MediaSense server will have the IP addresses of the Cisco
MediaSense nodes configured. The Unified Communications Manager SIP trunk
supports up to 16 destination IP addresses.

Note

Cisco
MediaSense cluster should have two or more nodes for redundancy and scalability
purposes.

Two topologies
are possible to set up Video on Hold:

Cisco
MediaSense is directly connected with the holding party's Unified
Communications Manager cluster.
When Video on
Hold server is located on the same cluster as the Holding Party, the SIP Trunk
on the Video on Hold server configuration should point to the Cisco MediaSense
server and the default content identifier should point to a stream ID that
exists on the MediaSense server. The content identifier can be any alphanumeric
string. No additional configuration is needed.

Cisco
MediaSenseis centrally deployed with Session Management Edition (SME).
When Video on
Hold server is located off the SME, the Video on Hold Server must be configured
on the leaf cluster hosting the Holding Party. The SIP trunk on this Video on
Hold Server should point to the SME SIP trunk. On the SME, a SIP Trunk should
be provisioned to point to the Cisco MediaSense server.
There are
three ways on how Unified Communications Manager can be configured to support
this centralized deployment:

Content
Identifier is numeric: In this case, a route pattern must be provisioned on the
SME to route the INVITE to Cisco MediaSense server. Essentially, we use the
left hand side of the INVITE URI sent by the leaf cluster to route the call to
Cisco MediaSense server. The right hand side of the INVITE URI contains the IP
address of SME node.

Content
Identifier is alpha-numeric and contains the IP address of Cisco MediaSense: In
this case, the content identifier configured on the leaf cluster should contain
both the stream-id and the IP address of the Cisco MediaSense server
(stream-id@mediasense-ipaddress).
On the SME
cluster, a SIP route pattern must be configured that uses the IP address
routing with the IP address of the MediaSense server matching the default
content identifier. In this scenario, the route is based on the right hand side
of the INVITE URI (IP address of MediaSense server) sent by the leaf cluster.

Content
Identifier is alpha-numeric and contains the domain name: In this case, the
content identifier configured on the leaf cluster should contain both the
stream-id and the domain name of the Cisco MediaSense server
(stream-id@cisco.com).
On the SME
cluster, a URI catalog must be created with a SIP route pattern as a Route
String. The content identifiers from the Cisco MediaSense server must be
imported into this catalog using the URI infrastructure.

Note

The SIP
Route Pattern must be configured to use domain routing. This SIP Route pattern
should point to a SIP Trunk to Cisco MediaSense server.

Video on Hold
SIP Trunk

The SIP Trunk
pointing to Video on Hold server should be configured with the default
configuration. The following information is needed for configuring the SIP
Trunk:

Name

Description

Device Pool

Location

Destination
address and destination port (multiple IP addresses and ports can be specified)

SIP Trunk
Security Profile

SIP Profile:
A SIP profile with the option ping configured should be selected. If none
exists, one should be created. This is not mandatory but will improve the user
experience.

Note

Other
configurations on the SIP Trunk are not supported for Video on Hold.

Cisco Unified
Communications Manager Administration Considerations

New configuration:
Video on Hold Server under Media Resources.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

There are two
performance counters that can be observed from RTMT for Video on Hold server:

Select the SIP Trunk to be used from the drop-down list. If a new SIP Trunk needs to be created, click the button Create SIP Trunk.

Step 7

Click Save.

Video QoS
Reservation

Cisco Unified
Communications Manager Administration Considerations

Note

This feature is
limited to use in lab environments for demonstration purposes only. Cisco
Technical Assistance Center (TAC) does not provide support for this feature.

The Video Quality
of Service (QoS) Reservation feature reserves bandwidth in a mobile network,
through a third party HTTP service, when a mobile device makes a call. This
reservation is only for VoIP calls made through
Cisco
Unified Communications Manager, not for other voice calls already
classified by the mobile network as voice calls.

For each device
with its MSISDN configured,
Unified Communications Manager requests its connection
type. If the connection type for the device is supported,
Unified Communications Manager reserves the bandwidth
with its MSISDN and the connected IP address. For a video call, there are two
separate reservations, one for the audio portion and one for the video portion,
both with the QoS Class Identifier (QCI) value set to 2. For an audio call,
there is one reservation, with the QCI value set to 1.

This feature only
supports
Unified Communications Manager SIP line side devices,
such as CSF client (Jabber for Tablet) and Cius.

To enable the
Video QoS feature, use the
System >
Service Parameters menu path to configure the parameters for the device. In
the
Clusterwide
Parameters section, configure
External QoS
Enabled to True.

To configure a
MSISDN for the device, use the
Device >
Phone menu path. Enter the MSISDN in the
Mobile
Subscriber ISDN(MSISDN) field.

Used in
conjunction with Web Service Root URI to query the device's connection type.

QoS URI

Used in
conjunction with Web Service Root URI to reserve bandwidth for the device.

Bulk
Administration Considerations

No changes.

CDR/CAR
Considerations

No changes.

IP Phones
Considerations

No changes.

RTMT
Considerations

No changes.

Security
Considerations

No changes.

Serviceability
Considerations

No changes.

Wireless LAN
Profiles

The Wireless LAN Profile feature removes the need for users to configure Wi-Fi parameters on their phones by allowing the administrator to configure Wi-Fi profiles for them. The user devices can automatically download the Wi-Fi configuration from the Cisco Unified Communications Manager TFTP server, and the configuration is then applied to these devices.

Before you create a Wireless LAN Profile, you can configure a Network Access Profile, which contains information about VPN connectivity and HTTP proxy settings. Create a Network Access Profile from the Device > Device Settings > Network Access Profile menu.

After you create one or more Wireless LAN Profiles, you can add them to a Wireless LAN Profile Group, which you can configure from the Device > Device Settings > Wireless LAN Profile Group menu. You can also specify the enterprise-wide default group.

Note

You may add up to four Wireless LAN Profiles to a Wireless LAN Profile Group.

Link Wireless LAN Profile Group with Device

You can link a Wireless LAN Profile Group at the device or device pool level.

Note

If you link a Wireless LAN Profile Group at the device and device pool level, Cisco Unified Communications Manager uses the device pool level.

Before You Begin

Create a Wireless LAN Profile Group.

Procedure

Step 1

Perform one of the following actions:

Select Device > Phone.

Select System > Device Pool

Step 2

Perform one of the following actions:

Find an existing device or create a new device.

Find an existing device pool or create a new device pool.

Step 3

Select a Wireless LAN Profile Group from the drop-down list box.

Step 4

Select Save.

The Wireless LAN Profile Group is linked to the device or Device Pool.

Wi-Fi Hotspot Profile

The Wi-Fi Hotspot Profile feature allows users to use their desk phones to provide a Wi-Fi Hotspot, so that they can connect a Wi-Fi device such as a tablet or a mobile phone to the network through the desk phone. The desk phones can automatically download the Wi-Fi Hotspot configuration from the Cisco Unified Communications Manager, and the configuration is then applied to these devices.

To use the Wi-Fi Hotspot Profile feature, you must configure a Wi-Fi Hotspot Profile on the Cisco Unified Communications Manager administrative interface. After the profile is created, you must associate it with a phone. To associate a Wi-Fi Hotspot Profile to a phone, you can configure the profile at the Enterprise Parameters, Common Phone Profile, or individual phone level. Configuring a Wi-Fi Hotspot Profile on the Phone page overrides the Enterprise Parameters and Common Phone Profile settings. After the desk phones download the TFTP configuration file, the users can enable Wi-Fi Hotspot and connect the Wi-Fi devices.

By default, the Wi-Fi Hotspot Profile feature is disabled in Cisco Unified Communications Manager. If you want to enable the Wi-Fi Hotspot for a desk phone, you can enable the Wi-Fi Hotspot feature at the Enterprise Phone Configuration, Common Phone Profile or individual phone level and then apply a Wi-Fi Hotspot Profile to the Enterprise Parameters, Common Phone Profile or individual phone level. The Wi-Fi Hotspot setting on the Phone page overrides the setting on the Common Phone Profile page, which overrides the setting on the Enterprise Phone Configuration page.

Cisco Unified Communications Manager Administration Considerations

The following new GUI menu path in Cisco Unified Communications Manager Administration allows you to configure Wi-Fi Hotspot Profile:

Device > Device Settings > Wi-Fi Hotspot Profile

The following GUI menu items have been updated for the Wi-Fi Hotspot Profile feature:

Wi-Fi Hotspot Profile Settings

Enter a name for the Wi-Fi Hotspot Profile. The value can include 1 to 50 characters, including alphanumeric characters, dots, dashes, and underscores.

Description

Enter a description for the Wi-Fi Hotspot Profile. The description can include up to 50 characters in any language, but it cannot include double quotation marks ("), percentage sign (%), ampersand (&), backslash (\), or angle brackets (<>).

User Modifiable

Select one of the following options from the drop-down list box:

Allowed

Indicates that the user can change any profile settings. This is the default setting.

Enter the Service Set Identifier (SSID) Prefix for the Wi-Fi Hotspot Profile. The SSID Prefix that you enter here is combined with the SSID suffix, which is generated automatically based on the local endpoint information, to create a unique SSID
for the Wi-Fi Hotspot of the phone.
The value can include 1 to 20 alphanumeric characters.

Frequency Band

Select one of the following frequency band settings from the drop-down list box:

Auto

The profile automatically chooses a frequency band.

2.4 GHz

The profile automatically chooses 2.4 GHz as the frequency band.

5 GHz

The profile automatically chooses 5 GHz as the frequency band.

Note

If you select the Auto option, a single channel will be used to serve clients because dual-band operation is currently not supported.

Authentication

Authentication Method

Specify the authentication method that is used to secure access to the Wi-Fi Hotspot. Depending on the method you choose, a PSK Passphrase, WEP key, or password description field appears so that you can provide the credentials that are required to connect to this Wi-Fi Hotspot.

If you choose this method, the Wi-Fi client that is connecting to the Wi-Fi Hotspot must be configured with a valid username and password.

PEAP-GTC

(Protected Extensible Authentication Protocol - Generic Token Card)

If you choose this method, the Wi-Fi client that is connecting to the Wi-Fi Hotspot must be configured with a valid username and password.

WPA2-PSK

(Wi-Fi Protected Access Pre-Shared Key)

This method uses Advanced Encryption Standard (AES) encryption. If you select this method, you must enter a passphrase, which is an 8 to 63 ASCII character string or a 64 HEX character string.

WPA-PSK

This method uses Temporal Key Integrity Protocol
(TKIP) encryption. If you select this method, you must enter a passphrase, which is an 8 to 63 ASCII character string or a 64 HEX character string.

WEP

(Wired Equivalent Privacy)

WEP requires a WEP Key, which is either a 5 or 13 ASCII character string or a 10 or 26 HEX character string.

None

No authentication is required.

Server Settings

Host Name/IP Address

Enter the DNS hostname (up to 255 characters) or IP address of the
authentication server.

Port

Enter the port number. 1812 is the default port. The accepted port range is 1-65535.

Shared Secret

Enter the shared secret. The value can include 1 to 32 characters.

The shared secret is used to authenticate against the
authentication server.
The shared secret specified in the Wi-Fi Hotspot Profile must match
with the shared secret specified in the authentication server.

Note

The server settings are displayed only if you select the authentication method as EAP-FAST, PEAP-MSCHAPv2, or PEAP-GTC.

Procedure changes

Create Wi-Fi Hotspot Profile

Use the following procedure to create a new Wi-Fi Hotspot Profile. After you create a Wi-Fi Hotspot Profile, you can apply it at the Enterprise Parameters, Common Phone Profile or individual
phone level.